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<p><a href="http://www.w3.org/"><img width="72" height="48" alt="W3C"
src="http://www.w3.org/Icons/w3c_home" /></a> </p>
<h1 id="title" class="title">Web Audio API </h1>
<h2 id="w3c-date-document"><acronym
title="World Wide Web Consortium">W3C</acronym> Editor's Draft
</h2>
<dl>
<dt>This version: </dt>
<dd><a
href="https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/specification.html">https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/specification.html</a>
</dd>
<dt>Latest published version: </dt>
<dd><a
href="http://www.w3.org/TR/webaudio/">http://www.w3.org/TR/webaudio/</a>
</dd>
<dt>Previous version: </dt>
<dd><a
href="http://www.w3.org/TR/2012/WD-webaudio-20120315/">http://www.w3.org/TR/2012/WD-webaudio-20120315/</a>
</dd>
</dl>
<dl>
<dt>Editor: </dt>
<dd>Chris Rogers, Google &lt;crogers@google.com&gt;</dd>
</dl>
<p class="copyright"><a
href="http://www.w3.org/Consortium/Legal/ipr-notice#Copyright">Copyright</a> ©
2012 <a href="http://www.w3.org/"><acronym
title="World Wide Web Consortium">W3C</acronym></a><sup>®</sup> (<a
href="http://www.csail.mit.edu/"><acronym
title="Massachusetts Institute of Technology">MIT</acronym></a>, <a
href="http://www.ercim.eu/"><acronym
title="European Research Consortium for Informatics and Mathematics">ERCIM</acronym></a>,
<a href="http://www.keio.ac.jp/">Keio</a>), All Rights Reserved. W3C <a
href="http://www.w3.org/Consortium/Legal/ipr-notice#Legal_Disclaimer">liability</a>,
<a
href="http://www.w3.org/Consortium/Legal/ipr-notice#W3C_Trademarks">trademark</a>
and <a href="http://www.w3.org/Consortium/Legal/copyright-documents">document
use</a> rules apply.</p>
<hr />
</div>
<div id="abstract-section" class="section">
<h2 id="abstract">Abstract</h2>
<p>This specification describes a high-level JavaScript <acronym
title="Application Programming Interface">API</acronym> for processing and
synthesizing audio in web applications. The primary paradigm is of an audio
routing graph, where a number of <a
href="#AudioNode-section"><code>AudioNode</code></a> objects are connected
together to define the overall audio rendering. The actual processing will
primarily take place in the underlying implementation (typically optimized
Assembly / C / C++ code), but <a href="#JavaScriptProcessing-section">direct
JavaScript processing and synthesis</a> is also supported. </p>
<p>The <a href="#introduction">introductory</a> section covers the motivation
behind this specification.</p>
<p>This API is designed to be used in conjunction with other APIs and elements
on the web platform, notably: XMLHttpRequest
(using the <code>responseType</code> and <code>response</code> attributes). For
games and interactive applications, it is anticipated to be used with the
<code>canvas</code> 2D and WebGL 3D graphics APIs. </p>
</div>
<div id="sotd-section" class="section">
<h2 id="sotd">Status of this Document</h2>
<p><em>This section describes the status of this document at the time of its
publication. Other documents may supersede this document. A list of current W3C
publications and the latest revision of this technical report can be found in
the <a href="http://www.w3.org/TR/">W3C technical reports index</a> at
http://www.w3.org/TR/. </em></p>
<p>This is the Editor's Draft of the <cite>Web Audio API</cite>
specification. It has been produced by the <a
href="http://www.w3.org/2011/audio/"><b>W3C Audio Working Group</b></a> , which
is part of the W3C WebApps Activity.</p>
<p></p>
<p>Please send comments about this document to &lt;<a
href="mailto:public-audio@w3.org">public-audio@w3.org</a>&gt; (<a
href="http://lists.w3.org/Archives/Public/public-audio/">public archives</a> of
the W3C audio mailing list). Web content and browser developers are encouraged
to review this draft. </p>
<p>Publication as a Working Draft does not imply endorsement by the W3C
Membership. This is a draft document and may be updated, replaced or obsoleted
by other documents at any time. It is inappropriate to cite this document as
other than work in progress.</p>
<p> This document was produced by a group operating under the <a href="http://www.w3.org/Consortium/Patent-Policy-20040205/">5 February 2004 W3C Patent Policy</a>. W3C maintains a <a rel="disclosure" href="http://www.w3.org/2004/01/pp-impl/46884/status">public list of any patent disclosures</a> made in connection with the deliverables of the group; that page also includes instructions for disclosing a patent. An individual who has actual knowledge of a patent which the individual believes contains <a href="http://www.w3.org/Consortium/Patent-Policy-20040205/#def-essential">Essential Claim(s)</a> must disclose the information in accordance with <a href="http://www.w3.org/Consortium/Patent-Policy-20040205/#sec-Disclosure">section 6 of the W3C Patent Policy</a>. </p>
</div>
<div id="toc">
<h2 id="L13522">Table of Contents</h2>
<div class="toc">
<ul>
<li><a href="#introduction">1. Introduction</a>
<ul>
<li><a href="#Features">1.1. Features</a></li>
<li><a href="#ModularRouting">1.2. Modular Routing</a></li>
<li><a href="#APIOverview">1.3. API Overview</a></li>
</ul>
</li>
<li><a href="#conformance">2. Conformance</a></li>
<li><a href="#API-section">4. The Audio API</a>
<ul>
<li><a href="#AudioContext-section">4.1. The AudioContext Interface</a>
<ul>
<li><a href="#attributes-AudioContext">4.1.1. Attributes</a></li>
<li><a href="#methodsandparams-AudioContext">4.1.2. Methods and
Parameters</a></li>
<li><a href="#lifetime-AudioContext">4.1.3. Lifetime</a></li>
</ul>
</li>
<li><a href="#OfflineAudioContext-section">4.1b. The OfflineAudioContext Interface</a>
</li>
<li><a href="#AudioNode-section">4.2. The AudioNode Interface</a>
<ul>
<li><a href="#attributes-AudioNode">4.2.1. Attributes</a></li>
<li><a href="#methodsandparams-AudioNode">4.2.2. Methods and
Parameters</a></li>
<li><a href="#lifetime-AudioNode">4.2.3. Lifetime</a></li>
</ul>
</li>
<li><a href="#AudioDestinationNode">4.4. The AudioDestinationNode
Interface</a>
<ul>
<li><a href="#attributes-AudioDestinationNode">4.4.1. Attributes</a></li>
</ul>
</li>
<li><a href="#AudioParam">4.5. The AudioParam Interface</a>
<ul>
<li><a href="#attributes-AudioParam">4.5.1. Attributes</a></li>
<li><a href="#methodsandparams-AudioParam">4.5.2. Methods and
Parameters</a></li>
<li><a href="#computedValue-AudioParam-section">4.5.3. Computation of Value</a></li>
<li><a href="#example1-AudioParam-section">4.5.4. AudioParam Automation Example</a></li>
</ul>
</li>
<li><a href="#GainNode">4.7. The GainNode Interface</a>
<ul>
<li><a href="#attributes-GainNode">4.7.1. Attributes</a></li>
</ul>
</li>
<li><a href="#DelayNode">4.8. The DelayNode Interface</a>
<ul>
<li><a href="#attributes-GainNode_2">4.8.1. Attributes</a></li>
</ul>
</li>
<li><a href="#AudioBuffer">4.9. The AudioBuffer Interface</a>
<ul>
<li><a href="#attributes-AudioBuffer">4.9.1. Attributes</a></li>
<li><a href="#methodsandparams-AudioBuffer">4.9.2. Methods and
Parameters</a></li>
</ul>
</li>
<li><a href="#AudioBufferSourceNode">4.10. The AudioBufferSourceNode
Interface</a>
<ul>
<li><a href="#attributes-AudioBufferSourceNode">4.10.1.
Attributes</a></li>
<li><a href="#methodsandparams-AudioBufferSourceNode">4.10.2. Methods and
Parameters</a></li>
</ul>
</li>
<li><a href="#MediaElementAudioSourceNode">4.11. The
MediaElementAudioSourceNode Interface</a></li>
<li><a href="#ScriptProcessorNode">4.12. The ScriptProcessorNode
Interface</a>
<ul>
<li><a href="#attributes-ScriptProcessorNode">4.12.1. Attributes</a></li>
</ul>
</li>
<li><a href="#AudioProcessingEvent">4.13. The AudioProcessingEvent
Interface</a>
<ul>
<li><a href="#attributes-AudioProcessingEvent">4.13.1. Attributes</a></li>
</ul>
</li>
<li><a href="#PannerNode">4.14. The PannerNode Interface</a>
<ul>
<li><a href="#attributes-PannerNode_attributes">4.14.2.
Attributes</a></li>
<li><a href="#Methods_and_Parameters">4.14.3. Methods and
Parameters</a></li>
</ul>
</li>
<li><a href="#AudioListener">4.15. The AudioListener Interface</a>
<ul>
<li><a href="#attributes-AudioListener">4.15.1. Attributes</a></li>
<li><a href="#L15842">4.15.2. Methods and Parameters</a></li>
</ul>
</li>
<li><a href="#ConvolverNode">4.16. The ConvolverNode Interface</a>
<ul>
<li><a href="#attributes-ConvolverNode">4.16.1. Attributes</a></li>
</ul>
</li>
<li><a href="#AnalyserNode">4.17. The AnalyserNode
Interface</a>
<ul>
<li><a href="#attributes-ConvolverNode_2">4.17.1. Attributes</a></li>
<li><a href="#methods-and-parameters">4.17.2. Methods and
Parameters</a></li>
</ul>
</li>
<li><a href="#ChannelSplitterNode">4.18. The ChannelSplitterNode
Interface</a>
<ul>
<li><a href="#example-1">Example:</a></li>
</ul>
</li>
<li><a href="#ChannelMergerNode">4.19. The ChannelMergerNode Interface</a>
<ul>
<li><a href="#example-2">Example:</a></li>
</ul>
</li>
<li><a href="#DynamicsCompressorNode">4.20. The DynamicsCompressorNode
Interface</a>
<ul>
<li><a href="#attributes-DynamicsCompressorNode">4.20.1.
Attributes</a></li>
</ul>
</li>
<li><a href="#BiquadFilterNode">4.21. The BiquadFilterNode Interface</a>
<ul>
<li><a href="#BiquadFilterNode-description">4.21.1 Lowpass</a></li>
<li><a href="#HIGHPASS">4.21.2 Highpass</a></li>
<li><a href="#BANDPASS">4.21.3 Bandpass</a></li>
<li><a href="#LOWSHELF">4.21.4 Lowshelf</a></li>
<li><a href="#L16352">4.21.5 Highshelf</a></li>
<li><a href="#PEAKING">4.21.6 Peaking</a></li>
<li><a href="#NOTCH">4.21.7 Notch</a></li>
<li><a href="#ALLPASS">4.21.8 Allpass</a></li>
<li><a href="#Methods">4.21.9. Methods</a></li>
</ul>
</li>
<li><a href="#WaveShaperNode">4.22. The WaveShaperNode Interface</a>
<ul>
<li><a href="#attributes-WaveShaperNode">4.22.1.
Attributes</a></li>
</ul>
</li>
<li><a href="#OscillatorNode">4.23. The OscillatorNode Interface</a>
<ul>
<li><a href="#attributes-OscillatorNode">4.23.1.
Attributes</a></li>
<li><a href="#methodsandparams-OscillatorNode-section">4.23.2. Methods and
Parameters</a></li>
</ul>
</li>
<li><a href="#PeriodicWave">4.24. The PeriodicWave Interface</a>
</li>
<li><a href="#MediaStreamAudioSourceNode">4.25. The
MediaStreamAudioSourceNode Interface</a></li>
<li><a href="#MediaStreamAudioDestinationNode">4.26. The
MediaStreamAudioDestinationNode Interface</a></li>
</ul>
</li>
<li><a href="#MixerGainStructure">6. Mixer Gain Structure</a>
<ul>
<li><a href="#background">Background</a></li>
<li><a href="#SummingJunction">Summing Inputs</a></li>
<li><a href="#gain-Control">Gain Control</a></li>
<li><a href="#Example-mixer-with-send-busses">Example: Mixer with Send
Busses</a></li>
</ul>
</li>
<li><a href="#DynamicLifetime">7. Dynamic Lifetime</a>
<ul>
<li><a href="#DynamicLifetime-background">Background</a></li>
<li><a href="#Example-DynamicLifetime">Example</a></li>
</ul>
</li>
<li><a href="#UpMix">9. Channel up-mixing and down-mixing</a>
<ul>
<li><a href="#ChannelLayouts">9.1. Speaker Channel Layouts</a>
<ul>
<li><a href="#ChannelOrdering">9.1.1. Channel Ordering</a></li>
<li><a href="#UpMix-sub">9.1.2. Up Mixing</a></li>
<li><a href="#down-mix">9.1.3. Down Mixing</a></li>
</ul>
</li>
<li><a href="#ChannelRules-section">9.2. Channel Rules Examples</a>
</ul>
</li>
<li><a href="#Spatialization">11. Spatialization / Panning </a>
<ul>
<li><a href="#Spatialization-background">Background</a></li>
<li><a href="#Spatialization-panning-algorithm">Panning Algorithm</a></li>
<li><a href="#Spatialization-distance-effects">Distance Effects</a></li>
<li><a href="#Spatialization-sound-cones">Sound Cones</a></li>
<li><a href="#Spatialization-doppler-shift">Doppler Shift</a></li>
</ul>
</li>
<li><a href="#Convolution">12. Linear Effects using Convolution</a>
<ul>
<li><a href="#Convolution-background">Background</a></li>
<li><a href="#Convolution-motivation">Motivation for use as a
Standard</a></li>
<li><a href="#Convolution-implementation-guide">Implementation Guide</a></li>
<li><a href="#Convolution-reverb-effect">Reverb Effect (with
matrixing)</a></li>
<li><a href="#recording-impulse-responses">Recording Impulse
Responses</a></li>
<li><a href="#tools">Tools</a></li>
<li><a href="#recording-setup">Recording Setup</a></li>
<li><a href="#warehouse">The Warehouse Space</a></li>
</ul>
</li>
<li><a href="#JavaScriptProcessing">13. JavaScript Synthesis and
Processing</a>
<ul>
<li><a href="#custom-DSP-effects">Custom DSP Effects</a></li>
<li><a href="#educational-applications">Educational Applications</a></li>
<li><a href="#javaScript-performance">JavaScript Performance</a></li>
</ul>
</li>
<li><a href="#Performance">15. Performance Considerations</a>
<ul>
<li><a href="#Latency">15.1. Latency: What it is and Why it's
Important</a></li>
<li><a href="#audio-glitching">15.2. Audio Glitching</a></li>
<li><a href="#hardware-scalability">15.3. Hardware Scalability</a>
<ul>
<li><a href="#CPU-monitoring">15.3.1. CPU monitoring</a></li>
<li><a href="#Voice-dropping">15.3.2. Voice Dropping</a></li>
<li><a href="#Simplification-of-Effects-Processing">15.3.3.
Simplification of Effects Processing</a></li>
<li><a href="#Sample-rate">15.3.4. Sample Rate</a></li>
<li><a href="#pre-flighting">15.3.5. Pre-flighting</a></li>
<li><a href="#Authoring-for-different-user-agents">15.3.6. Authoring
for different user agents</a></li>
<li><a href="#Scalability-of-Direct-JavaScript-Synthesis">15.3.7.
Scalability of Direct JavaScript Synthesis / Processing</a></li>
</ul>
</li>
<li><a href="#JavaScriptPerformance">15.4. JavaScript Issues with
real-time Processing and Synthesis: </a></li>
</ul>
</li>
<li><a href="#ExampleApplications">16. Example Applications</a>
<ul>
<li><a href="#basic-sound-playback">Basic Sound Playback</a></li>
<li><a href="#threeD-environmentse-and-games">3D Environments and
Games</a></li>
<li><a href="#musical-applications">Musical Applications</a></li>
<li><a href="#music-visualizers">Music Visualizers</a></li>
<li><a href="#educational-applications_2">Educational
Applications</a></li>
<li><a href="#artistic-audio-exploration">Artistic Audio
Exploration</a></li>
</ul>
</li>
<li><a href="#SecurityConsiderations">17. Security Considerations</a></li>
<li><a href="#PrivacyConsiderations">18. Privacy Considerations</a></li>
<li><a href="#requirements">19. Requirements and Use Cases</a></li>
<li><a href="#OldNames">20. Old Names</a></li>
<li><a href="#L17310">A.References</a>
<ul>
<li><a href="#Normative-references">A.1 Normative references</a></li>
<li><a href="#Informative-references">A.2 Informative references</a></li>
</ul>
</li>
<li><a href="#L17335">B.Acknowledgements</a></li>
<li><a href="#ChangeLog">C. Web Audio API Change Log</a></li>
</ul>
</div>
</div>
<div id="sections">
<div id="div-introduction" class="section">
<h2 id="introduction">1. Introduction</h2>
<p class="norm">This section is informative.</p>
<p>Audio on the web has been fairly primitive up to this point and until very
recently has had to be delivered through plugins such as Flash and QuickTime.
The introduction of the <code>audio</code> element in HTML5 is very important,
allowing for basic streaming audio playback. But, it is not powerful enough to
handle more complex audio applications. For sophisticated web-based games or
interactive applications, another solution is required. It is a goal of this
specification to include the capabilities found in modern game audio engines as
well as some of the mixing, processing, and filtering tasks that are found in
modern desktop audio production applications. </p>
<p>The APIs have been designed with a wide variety of <a
href="#ExampleApplications-section">use cases</a> in mind. Ideally, it should
be able to support <i>any</i> use case which could reasonably be implemented
with an optimized C++ engine controlled via JavaScript and run in a browser.
That said, modern desktop audio software can have very advanced capabilities,
some of which would be difficult or impossible to build with this system.
Apple's Logic Audio is one such application which has support for external MIDI
controllers, arbitrary plugin audio effects and synthesizers, highly optimized
direct-to-disk audio file reading/writing, tightly integrated time-stretching,
and so on. Nevertheless, the proposed system will be quite capable of
supporting a large range of reasonably complex games and interactive
applications, including musical ones. And it can be a very good complement to
the more advanced graphics features offered by WebGL. The API has been designed
so that more advanced capabilities can be added at a later time. </p>
<div id="Features-section" class="section">
<h2 id="Features">1.1. Features</h2>
</div>
<p>The API supports these primary features: </p>
<ul>
<li><a href="#ModularRouting-section">Modular routing</a> for simple or
complex mixing/effect architectures, including <a
href="#MixerGainStructure-section">multiple sends and submixes</a>.</li>
<li><a href="#AudioParam">Sample-accurate scheduled sound
playback</a> with low <a href="#Latency-section">latency</a> for musical
applications requiring a very high degree of rhythmic precision such as
drum machines and sequencers. This also includes the possibility of <a
href="#DynamicLifetime-section">dynamic creation</a> of effects. </li>
<li>Automation of audio parameters for envelopes, fade-ins / fade-outs,
granular effects, filter sweeps, LFOs etc. </li>
<li>Flexible handling of channels in an audio stream, allowing them to be split and merged.</li>
<li>Processing of audio sources from an <code>audio</code> or
<code>video</code> <a href="#MediaElementAudioSourceNode">media
element</a>. </li>
<li>Processing live audio input using a <a href="#MediaStreamAudioSourceNode">MediaStream</a>
from getUserMedia().
</li>
<li>Integration with WebRTC
<ul>
<li>Processing audio received from a remote peer using a <a href="#MediaStreamAudioSourceNode">MediaStream</a>.
</li>
<li>Sending a generated or processed audio stream to a remote peer using a <a href="#MediaStreamAudioDestinationNode">MediaStream</a>.
</li>
</ul>
</li>
<li>Audio stream synthesis and processing <a
href="#JavaScriptProcessing-section">directly in JavaScript</a>. </li>
<li><a href="#Spatialization-section">Spatialized audio</a> supporting a wide
range of 3D games and immersive environments:
<ul>
<li>Panning models: equal-power, HRTF, pass-through </li>
<li>Distance Attenuation </li>
<li>Sound Cones </li>
<li>Obstruction / Occlusion </li>
<li>Doppler Shift </li>
<li>Source / Listener based</li>
</ul>
</li>
<li>A <a href="#Convolution-section">convolution engine</a> for a wide range
of linear effects, especially very high-quality room effects. Here are some
examples of possible effects:
<ul>
<li>Small / large room </li>
<li>Cathedral </li>
<li>Concert hall </li>
<li>Cave </li>
<li>Tunnel </li>
<li>Hallway </li>
<li>Forest </li>
<li>Amphitheater </li>
<li>Sound of a distant room through a doorway </li>
<li>Extreme filters</li>
<li>Strange backwards effects</li>
<li>Extreme comb filter effects </li>
</ul>
</li>
<li>Dynamics compression for overall control and sweetening of the mix </li>
<li>Efficient <a href="#AnalyserNode">real-time time-domain and
frequency analysis / music visualizer support</a></li>
<li>Efficient biquad filters for lowpass, highpass, and other common filters.
</li>
<li>A Waveshaping effect for distortion and other non-linear effects</li>
<li>Oscillators</li>
</ul>
<div id="ModularRouting-section">
<h2 id="ModularRouting">1.2. Modular Routing</h2>
<p>Modular routing allows arbitrary connections between different <a
href="#AudioNode-section"><code>AudioNode</code></a> objects. Each node can
have <dfn>inputs</dfn> and/or <dfn>outputs</dfn>. A <dfn>source node</dfn> has no inputs
and a single output. A <dfn>destination node</dfn> has
one input and no outputs, the most common example being <a
href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a> the final destination to the audio
hardware. Other nodes such as filters can be placed between the source and destination nodes.
The developer doesn't have to worry about low-level stream format details
when two objects are connected together; <a href="#UpMix-section">the right
thing just happens</a>. For example, if a mono audio stream is connected to a
stereo input it should just mix to left and right channels <a
href="#UpMix-section">appropriately</a>. </p>
<p>In the simplest case, a single source can be routed directly to the output.
All routing occurs within an <a
href="#AudioContext-section"><code>AudioContext</code></a> containing a single
<a href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a>:
</p>
<img alt="modular routing" src="images/modular-routing1.png" />
<p>Illustrating this simple routing, here's a simple example playing a single
sound: </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">ECMAScript</span> </div>
<div class="blockContent">
<pre class="code"><code class="es-code">
var context = new AudioContext();
function playSound() {
var source = context.createBufferSource();
source.buffer = dogBarkingBuffer;
source.connect(context.destination);
source.start(0);
}
</code></pre>
</div>
</div>
<p>Here's a more complex example with three sources and a convolution reverb
send with a dynamics compressor at the final output stage: </p>
<img alt="modular routing2" src="images/modular-routing2.png" />
<div class="example">
<div class="exampleHeader">
Example</div>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">ECMAScript</span></div>
<div class="blockContent">
<pre class="code"><code class="es-code">
var context = 0;
var compressor = 0;
var reverb = 0;
var source1 = 0;
var source2 = 0;
var source3 = 0;
var lowpassFilter = 0;
var waveShaper = 0;
var panner = 0;
var dry1 = 0;
var dry2 = 0;
var dry3 = 0;
var wet1 = 0;
var wet2 = 0;
var wet3 = 0;
var masterDry = 0;
var masterWet = 0;
function setupRoutingGraph () {
context = new AudioContext();
// Create the effects nodes.
lowpassFilter = context.createBiquadFilter();
waveShaper = context.createWaveShaper();
panner = context.createPanner();
compressor = context.createDynamicsCompressor();
reverb = context.createConvolver();
// Create master wet and dry.
masterDry = context.createGain();
masterWet = context.createGain();
// Connect final compressor to final destination.
compressor.connect(context.destination);
// Connect master dry and wet to compressor.
masterDry.connect(compressor);
masterWet.connect(compressor);
// Connect reverb to master wet.
reverb.connect(masterWet);
// Create a few sources.
source1 = context.createBufferSource();
source2 = context.createBufferSource();
source3 = context.createOscillator();
source1.buffer = manTalkingBuffer;
source2.buffer = footstepsBuffer;
source3.frequency.value = 440;
// Connect source1
dry1 = context.createGain();
wet1 = context.createGain();
source1.connect(lowpassFilter);
lowpassFilter.connect(dry1);
lowpassFilter.connect(wet1);
dry1.connect(masterDry);
wet1.connect(reverb);
// Connect source2
dry2 = context.createGain();
wet2 = context.createGain();
source2.connect(waveShaper);
waveShaper.connect(dry2);
waveShaper.connect(wet2);
dry2.connect(masterDry);
wet2.connect(reverb);
// Connect source3
dry3 = context.createGain();
wet3 = context.createGain();
source3.connect(panner);
panner.connect(dry3);
panner.connect(wet3);
dry3.connect(masterDry);
wet3.connect(reverb);
// Start the sources now.
source1.start(0);
source2.start(0);
source3.start(0);
}
</code></pre>
</div>
</div>
</div>
</div>
</div>
<div id="APIOverview-section" class="section">
<h2 id="APIOverview">1.3. API Overview</h2>
</div>
<p>The interfaces defined are: </p>
<ul>
<li>An <a class="dfnref" href="#AudioContext-section">AudioContext</a>
interface, which contains an audio signal graph representing connections
betweens AudioNodes. </li>
<li>An <a class="dfnref" href="#AudioNode-section">AudioNode</a> interface,
which represents audio sources, audio outputs, and intermediate processing
modules. AudioNodes can be dynamically connected together in a <a
href="#ModularRouting-section">modular fashion</a>. <code>AudioNodes</code>
exist in the context of an <code>AudioContext</code> </li>
<li>An <a class="dfnref"
href="#AudioDestinationNode-section">AudioDestinationNode</a> interface, an
AudioNode subclass representing the final destination for all rendered
audio. </li>
<li>An <a class="dfnref" href="#AudioBuffer-section">AudioBuffer</a>
interface, for working with memory-resident audio assets. These can
represent one-shot sounds, or longer audio clips. </li>
<li>An <a class="dfnref"
href="#AudioBufferSourceNode-section">AudioBufferSourceNode</a> interface,
an AudioNode which generates audio from an AudioBuffer. </li>
<li>A <a class="dfnref"
href="#MediaElementAudioSourceNode-section">MediaElementAudioSourceNode</a>
interface, an AudioNode which is the audio source from an
<code>audio</code>, <code>video</code>, or other media element. </li>
<li>A <a class="dfnref"
href="#MediaStreamAudioSourceNode-section">MediaStreamAudioSourceNode</a>
interface, an AudioNode which is the audio source from a
MediaStream such as live audio input, or from a remote peer. </li>
<li>A <a class="dfnref"
href="#MediaStreamAudioDestinationNode-section">MediaStreamAudioDestinationNode</a>
interface, an AudioNode which is the audio destination to a
MediaStream sent to a remote peer. </li>
<li>A <a class="dfnref"
href="#ScriptProcessorNode-section">ScriptProcessorNode</a> interface, an
AudioNode for generating or processing audio directly in JavaScript. </li>
<li>An <a class="dfnref"
href="#AudioProcessingEvent-section">AudioProcessingEvent</a> interface,
which is an event type used with <code>ScriptProcessorNode</code> objects.
</li>
<li>An <a class="dfnref" href="#AudioParam-section">AudioParam</a> interface,
for controlling an individual aspect of an AudioNode's functioning, such as
volume. </li>
<li>An <a class="dfnref" href="#GainNode-section">GainNode</a>
interface, for explicit gain control. Because inputs to AudioNodes support
multiple connections (as a unity-gain summing junction), mixers can be <a
href="#MixerGainStructure-section">easily built</a> with GainNodes.
</li>
<li>A <a class="dfnref" href="#BiquadFilterNode-section">BiquadFilterNode</a>
interface, an AudioNode for common low-order filters such as:
<ul>
<li>Low Pass</li>
<li>High Pass </li>
<li>Band Pass </li>
<li>Low Shelf </li>
<li>High Shelf </li>
<li>Peaking </li>
<li>Notch </li>
<li>Allpass </li>
</ul>
</li>
<li>A <a class="dfnref" href="#DelayNode-section">DelayNode</a> interface, an
AudioNode which applies a dynamically adjustable variable delay. </li>
<li>An <a class="dfnref" href="#PannerNode-section">PannerNode</a>
interface, for spatializing / positioning audio in 3D space. </li>
<li>An <a class="dfnref" href="#AudioListener-section">AudioListener</a>
interface, which works with an <code>PannerNode</code> for
spatialization. </li>
<li>A <a class="dfnref" href="#ConvolverNode-section">ConvolverNode</a>
interface, an AudioNode for applying a <a
href="#Convolution-section">real-time linear effect</a> (such as the sound
of a concert hall). </li>
<li>A <a class="dfnref"
href="#AnalyserNode-section">AnalyserNode</a> interface,
for use with music visualizers, or other visualization applications. </li>
<li>A <a class="dfnref"
href="#ChannelSplitterNode-section">ChannelSplitterNode</a> interface,
for accessing the individual channels of an audio stream in the routing
graph. </li>
<li>A <a class="dfnref"
href="#ChannelMergerNode-section">ChannelMergerNode</a> interface, for
combining channels from multiple audio streams into a single audio stream.
</li>
<li>A <a
href="#DynamicsCompressorNode-section">DynamicsCompressorNode</a> interface, an
AudioNode for dynamics compression. </li>
<li>A <a class="dfnref" href="#dfn-WaveShaperNode">WaveShaperNode</a>
interface, an AudioNode which applies a non-linear waveshaping effect for
distortion and other more subtle warming effects. </li>
<li>A <a class="dfnref" href="#dfn-OscillatorNode">OscillatorNode</a>
interface, an audio source generating a periodic waveform. </li>
</ul>
</div>
<div id="conformance-section" class="section">
<h2 id="conformance">2. Conformance</h2>
<p>Everything in this specification is normative except for examples and
sections marked as being informative. </p>
<p>The keywords “<span class="rfc2119">MUST</span>”, “<span
class="rfc2119">MUST NOT</span>”, “<span
class="rfc2119">REQUIRED</span>”, “<span class="rfc2119">SHALL</span>”,
“<span class="rfc2119">SHALL NOT</span>”, “<span
class="rfc2119">RECOMMENDED</span>”, “<span class="rfc2119">MAY</span>”
and “<span class="rfc2119">OPTIONAL</span>” in this document are to be
interpreted as described in <cite><a href="http://www.ietf.org/rfc/rfc2119">Key
words for use in RFCs to Indicate Requirement Levels</a></cite> <a
href="#RFC2119">[RFC2119]</a>. </p>
<p>The following conformance classes are defined by this specification: </p>
<dl>
<dt><dfn id="dfn-conforming-implementation">conforming
implementation</dfn></dt>
<dd><p>A user agent is considered to be a <a class="dfnref"
href="#dfn-conforming-implementation">conforming implementation</a> if it
satisfies all of the <span class="rfc2119">MUST</span>-, <span
class="rfc2119">REQUIRED</span>- and <span
class="rfc2119">SHALL</span>-level criteria in this specification that
apply to implementations. </p>
</dd>
</dl>
</div>
<div id="terminology-section" class="section">
<div id="API-section-section" class="section">
<h2 id="API-section">4. The Audio API</h2>
</div>
<div id="AudioContext-section-section" class="section">
<h2 id="AudioContext-section">4.1. The AudioContext Interface</h2>
<p>This interface represents a set of <a
href="#AudioNode-section"><code>AudioNode</code></a> objects and their
connections. It allows for arbitrary routing of signals to the <a
href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a>
(what the user ultimately hears). Nodes are created from the context and are
then <a href="#ModularRouting-section">connected</a> together. In most use
cases, only a single AudioContext is used per document.</p>
<br>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-context-idl">
callback DecodeSuccessCallback = void (AudioBuffer decodedData);
callback DecodeErrorCallback = void ();
[Constructor]
interface <dfn id="dfn-AudioContext">AudioContext</dfn> : EventTarget {
readonly attribute AudioDestinationNode destination;
readonly attribute float sampleRate;
readonly attribute double currentTime;
readonly attribute AudioListener listener;
AudioBuffer createBuffer(unsigned long numberOfChannels, unsigned long length, float sampleRate);
void decodeAudioData(ArrayBuffer audioData,
DecodeSuccessCallback successCallback,
optional DecodeErrorCallback errorCallback);
<span class="comment">// AudioNode creation </span>
AudioBufferSourceNode createBufferSource();
MediaElementAudioSourceNode createMediaElementSource(HTMLMediaElement mediaElement);
MediaStreamAudioSourceNode createMediaStreamSource(MediaStream mediaStream);
MediaStreamAudioDestinationNode createMediaStreamDestination();
ScriptProcessorNode createScriptProcessor(optional unsigned long bufferSize = 0,
optional unsigned long numberOfInputChannels = 2,
optional unsigned long numberOfOutputChannels = 2);
AnalyserNode createAnalyser();
GainNode createGain();
DelayNode createDelay(optional double maxDelayTime = 1.0);
BiquadFilterNode createBiquadFilter();
WaveShaperNode createWaveShaper();
PannerNode createPanner();
ConvolverNode createConvolver();
ChannelSplitterNode createChannelSplitter(optional unsigned long numberOfOutputs = 6);
ChannelMergerNode createChannelMerger(optional unsigned long numberOfInputs = 6);
DynamicsCompressorNode createDynamicsCompressor();
OscillatorNode createOscillator();
PeriodicWave createPeriodicWave(Float32Array real, Float32Array imag);
};
</code></pre>
</div>
</div>
<div id="attributes-AudioContext-section" class="section">
<h3 id="attributes-AudioContext">4.1.1. Attributes</h3>
<dl>
<dt id="dfn-destination"><code>destination</code></dt>
<dd><p>An <a
href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a>
with a single input representing the final destination for all audio.
Usually this will represent the actual audio hardware.
All AudioNodes actively rendering
audio will directly or indirectly connect to <code>destination</code>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-sampleRate"><code>sampleRate</code></dt>
<dd><p>The sample rate (in sample-frames per second) at which the
AudioContext handles audio. It is assumed that all AudioNodes in the
context run at this rate. In making this assumption, sample-rate
converters or "varispeed" processors are not supported in real-time
processing.</p>
</dd>
</dl>
<dl>
<dt id="dfn-currentTime"><code>currentTime</code></dt>
<dd><p>This is a time in seconds which starts at zero when the context is
created and increases in real-time. All scheduled times are relative to
it. This is not a "transport" time which can be started, paused, and
re-positioned. It is always moving forward. A GarageBand-like timeline
transport system can be very easily built on top of this (in JavaScript).
This time corresponds to an ever-increasing hardware timestamp. </p>
</dd>
</dl>
<dl>
<dt id="dfn-listener"><code>listener</code></dt>
<dd><p>An <a href="#AudioListener-section"><code>AudioListener</code></a>
which is used for 3D <a
href="#Spatialization-section">spatialization</a>.</p>
</dd>
</dl>
</div>
<div id="methodsandparams-AudioContext-section" class="section">
<h3 id="methodsandparams-AudioContext">4.1.2. Methods and Parameters</h3>
<dl>
<dt id="dfn-createBuffer">The <code>createBuffer</code> method</dt>
<dd><p>Creates an AudioBuffer of the given size. The audio data in the
buffer will be zero-initialized (silent). An NOT_SUPPORTED_ERR exception will be thrown if
the <code>numberOfChannels</code> or <code>sampleRate</code> are out-of-bounds,
or if length is 0.</p>
<p>The <dfn id="dfn-numberOfChannels">numberOfChannels</dfn> parameter
determines how many channels the buffer will have. An implementation must support at least 32 channels. </p>
<p>The <dfn id="dfn-length">length</dfn> parameter determines the size of
the buffer in sample-frames. </p>
<p>The <dfn id="dfn-sampleRate_2">sampleRate</dfn> parameter describes
the sample-rate of the linear PCM audio data in the buffer in
sample-frames per second. An implementation must support sample-rates in at least the range 22050 to 96000.</p>
</dd>
</dl>
<dl>
<dt id="dfn-decodeAudioData">The <code>decodeAudioData</code> method</dt>
<dd><p>Asynchronously decodes the audio file data contained in the
ArrayBuffer. The ArrayBuffer can, for example, be loaded from an XMLHttpRequest's
<code>response</code> attribute after setting the <code>responseType</code> to "arraybuffer".
Audio file data can be in any of the
formats supported by the <code>audio</code> element. </p>
<p><dfn id="dfn-audioData">audioData</dfn> is an ArrayBuffer containing
audio file data.</p>
<p><dfn id="dfn-successCallback">successCallback</dfn> is a callback
function which will be invoked when the decoding is finished. The single
argument to this callback is an AudioBuffer representing the decoded PCM
audio data.</p>
<p><dfn id="dfn-errorCallback">errorCallback</dfn> is a callback function
which will be invoked if there is an error decoding the audio file
data.</p>
<p>
The following steps must be performed:
</p>
<ol>
<li>Temporarily neuter the <dfn>audioData</dfn> ArrayBuffer in such a way that JavaScript code may not
access or modify the data.</li>
<li>Queue a decoding operation to be performed on another thread.</li>
<li>The decoding thread will attempt to decode the encoded <dfn>audioData</dfn> into linear PCM.
If a decoding error is encountered due to the audio format not being recognized or supported, or
because of corrupted/unexpected/inconsistent data then the <dfn>audioData</dfn> neutered state
will be restored to normal and the <dfn>errorCallback</dfn> will be
scheduled to run on the main thread's event loop and these steps will be terminated.</li>
<li>The decoding thread will take the result, representing the decoded linear PCM audio data,
and resample it to the sample-rate of the AudioContext if it is different from the sample-rate
of <dfn>audioData</dfn>. The final result (after possibly sample-rate converting) will be stored
in an AudioBuffer.
</li>
<li>The <dfn>audioData</dfn> neutered state will be restored to normal
</li>
<li>
The <dfn>successCallback</dfn> function will be scheduled to run on the main thread's event loop
given the AudioBuffer from step (4) as an argument.
</li>
</ol>
</dd>
</dl>
<dl>
<dt id="dfn-createBufferSource">The <code>createBufferSource</code>
method</dt>
<dd><p>Creates an <a
href="#AudioBufferSourceNode-section"><code>AudioBufferSourceNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createMediaElementSource">The <code>createMediaElementSource</code>
method</dt>
<dd><p>Creates a <a
href="#MediaElementAudioSourceNode-section"><code>MediaElementAudioSourceNode</code></a> given an HTMLMediaElement.
As a consequence of calling this method, audio playback from the HTMLMediaElement will be re-routed
into the processing graph of the AudioContext.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createMediaStreamSource">The <code>createMediaStreamSource</code>
method</dt>
<dd><p>Creates a <a
href="#MediaStreamAudioSourceNode-section"><code>MediaStreamAudioSourceNode</code></a> given a MediaStream.
As a consequence of calling this method, audio playback from the MediaStream will be re-routed
into the processing graph of the AudioContext.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createMediaStreamDestination">The <code>createMediaStreamDestination</code>
method</dt>
<dd><p>Creates a <a
href="#MediaStreamAudioDestinationNode-section"><code>MediaStreamAudioDestinationNode</code></a>.
</p>
</dd>
</dl>
<dl>
<dt id="dfn-createScriptProcessor">The <code>createScriptProcessor</code>
method</dt>
<dd><p>Creates a <a
href="#ScriptProcessorNode"><code>ScriptProcessorNode</code></a> for
direct audio processing using JavaScript. An INDEX_SIZE_ERR exception MUST be thrown if <code>bufferSize</code> or <code>numberOfInputChannels</code> or <code>numberOfOutputChannels</code>
are outside the valid range. </p>
<p>The <dfn id="dfn-bufferSize">bufferSize</dfn> parameter determines the
buffer size in units of sample-frames. If it's not passed in, or if the
value is 0, then the implementation will choose the best buffer size for
the given environment, which will be constant power of 2 throughout the lifetime
of the node. Otherwise if the author explicitly specifies the bufferSize,
it must be one of the following values: 256, 512, 1024, 2048, 4096, 8192,
16384. This value controls how
frequently the <code>audioprocess</code> event is dispatched and
how many sample-frames need to be processed each call. Lower values for
<code>bufferSize</code> will result in a lower (better) <a
href="#Latency-section">latency</a>. Higher values will be necessary to
avoid audio breakup and <a href="#Glitching-section">glitches</a>.
It is recommended for authors to not specify this buffer size and allow
the implementation to pick a good buffer size to balance between latency
and audio quality.
</p>
<p>The <dfn id="dfn-numberOfInputChannels">numberOfInputChannels</dfn> parameter (defaults to 2) and
determines the number of channels for this node's input. Values of up to 32 must be supported. </p>
<p>The <dfn id="dfn-numberOfOutputChannels">numberOfOutputChannels</dfn> parameter (defaults to 2) and
determines the number of channels for this node's output. Values of up to 32 must be supported.</p>
<p>It is invalid for both <code>numberOfInputChannels</code> and
<code>numberOfOutputChannels</code> to be zero. </p>
</dd>
</dl>
<dl>
<dt id="dfn-createAnalyser">The <code>createAnalyser</code> method</dt>
<dd><p>Creates a <a
href="#AnalyserNode-section"><code>AnalyserNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createGain">The <code>createGain</code> method</dt>
<dd><p>Creates a <a
href="#GainNode-section"><code>GainNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createDelay">The <code>createDelay</code> method</dt>
<dd><p>Creates a <a href="#DelayNode-section"><code>DelayNode</code></a>
representing a variable delay line. The initial default delay time will
be 0 seconds.</p>
<p>The <dfn id="dfn-maxDelayTime">maxDelayTime</dfn> parameter is
optional and specifies the maximum delay time in seconds allowed for the delay line. If specified, this value MUST be
greater than zero and less than three minutes or a NOT_SUPPORTED_ERR exception will be thrown.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createBiquadFilter">The <code>createBiquadFilter</code>
method</dt>
<dd><p>Creates a <a
href="#BiquadFilterNode-section"><code>BiquadFilterNode</code></a>
representing a second order filter which can be configured as one of
several common filter types.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createWaveShaper">The <code>createWaveShaper</code>
method</dt>
<dd><p>Creates a <a
href="#WaveShaperNode-section"><code>WaveShaperNode</code></a>
representing a non-linear distortion.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createPanner">The <code>createPanner</code> method</dt>
<dd><p>Creates an <a
href="#PannerNode-section"><code>PannerNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createConvolver">The <code>createConvolver</code> method</dt>
<dd><p>Creates a <a
href="#ConvolverNode-section"><code>ConvolverNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createChannelSplitter">The <code>createChannelSplitter</code>
method</dt>
<dd><p>Creates an <a
href="#ChannelSplitterNode-section"><code>ChannelSplitterNode</code></a>
representing a channel splitter. An exception will be thrown for invalid parameter values.</p>
<p>The <dfn id="dfn-numberOfOutputs">numberOfOutputs</dfn> parameter
determines the number of outputs. Values of up to 32 must be supported. If not specified, then 6 will be used. </p>
</dd>
</dl>
<dl>
<dt id="dfn-createChannelMerger">The <code>createChannelMerger</code>
method</dt>
<dd><p>Creates an <a
href="#ChannelMergerNode-section"><code>ChannelMergerNode</code></a>
representing a channel merger. An exception will be thrown for invalid parameter values.</p>
<p>The <dfn id="dfn-numberOfInputs">numberOfInputs</dfn> parameter
determines the number of inputs. Values of up to 32 must be supported. If not specified, then 6 will be used. </p>
</dd>
</dl>
<dl>
<dt id="dfn-createDynamicsCompressor">The
<code>createDynamicsCompressor</code> method</dt>
<dd><p>Creates a <a
href="#DynamicsCompressorNode-section"><code>DynamicsCompressorNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createOscillator">The
<code>createOscillator</code> method</dt>
<dd><p>Creates an <a
href="#OscillatorNode-section"><code>OscillatorNode</code></a>.</p>
</dd>
</dl>
<dl>
<dt id="dfn-createPeriodicWave">The
<code>createPeriodicWave</code> method</dt>
<dd><p>Creates a <a
href="#PeriodicWave-section"><code>PeriodicWave</code></a> representing a waveform containing arbitrary harmonic content.
The <code>real</code> and <code>imag</code> parameters must be of type <code>Float32Array</code> of equal
lengths greater than zero and less than or equal to 4096 or an exception will be thrown.
These parameters specify the Fourier coefficients of a
<a href="http://en.wikipedia.org/wiki/Fourier_series">Fourier series</a> representing the partials of a periodic waveform.
The created PeriodicWave will be used with an <a href="#OscillatorNode-section"><code>OscillatorNode</code></a>
and will represent a <em>normalized</em> time-domain waveform having maximum absolute peak value of 1.
Another way of saying this is that the generated waveform of an <a href="#OscillatorNode-section"><code>OscillatorNode</code></a>
will have maximum peak value at 0dBFS. Conveniently, this corresponds to the full-range of the signal values used by the Web Audio API.
Because the PeriodicWave will be normalized on creation, the <code>real</code> and <code>imag</code> parameters
represent <em>relative</em> values.
</p>
<p>The <dfn id="dfn-real">real</dfn> parameter represents an array of <code>cosine</code> terms (traditionally the A terms).
In audio terminology, the first element (index 0) is the DC-offset of the periodic waveform and is usually set to zero.
The second element (index 1) represents the fundamental frequency. The third element represents the first overtone, and so on.</p>
<p>The <dfn id="dfn-imag">imag</dfn> parameter represents an array of <code>sine</code> terms (traditionally the B terms).
The first element (index 0) should be set to zero (and will be ignored) since this term does not exist in the Fourier series.
The second element (index 1) represents the fundamental frequency. The third element represents the first overtone, and so on.</p>
</dd>
</dl>
</div>
</div>
<h3 id="lifetime-AudioContext">4.1.3. Lifetime</h3>
<p class="norm">This section is informative.</p>
<p>
Once created, an <code>AudioContext</code> will continue to play sound until it has no more sound to play, or
the page goes away.
</p>
<div id="OfflineAudioContext-section-section" class="section">
<h2 id="OfflineAudioContext-section">4.1b. The OfflineAudioContext Interface</h2>
<p>
OfflineAudioContext is a particular type of AudioContext for rendering/mixing-down (potentially) faster than real-time.
It does not render to the audio hardware, but instead renders as quickly as possible, calling a completion event handler
with the result provided as an AudioBuffer.
</p>
<p>
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="offline-audio-context-idl">
[Constructor(unsigned long numberOfChannels, unsigned long length, float sampleRate)]
interface <dfn id="dfn-OfflineAudioContext">OfflineAudioContext</dfn> : AudioContext {
void startRendering();
attribute EventHandler oncomplete;
};
</code></pre>
</div>
</div>
<div id="attributes-OfflineAudioContext-section" class="section">
<h3 id="attributes-OfflineAudioContext">4.1b.1. Attributes</h3>
<dl>
<dt id="dfn-oncomplete"><code>oncomplete</code></dt>
<dd><p>An EventHandler of type <a href="#OfflineAudioCompletionEvent-section">OfflineAudioCompletionEvent</a>.</p>
</dd>
</dl>
</div>
<div id="methodsandparams-OfflineAudioContext-section" class="section">
<h3 id="methodsandparams-OfflineAudioContext">4.1b.2. Methods and Parameters</h3>
<dl>
<dt id="dfn-startRendering">The <code>startRendering</code>
method</dt>
<dd><p>Given the current connections and scheduled changes, starts rendering audio. The
<code>oncomplete</code> handler will be called once the rendering has finished.
This method must only be called one time or an exception will be thrown.</p>
</dd>
</dl>
</div>
<div id="OfflineAudioCompletionEvent-section" class="section">
<h2 id="OfflineAudioCompletionEvent">4.1c. The OfflineAudioCompletionEvent Interface</h2>
<p>This is an <code>Event</code> object which is dispatched to <a
href="#OfflineAudioContext-section"><code>OfflineAudioContext</code></a>. </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="offline-audio-completion-event-idl">
interface <dfn id="dfn-OfflineAudioCompletionEvent">OfflineAudioCompletionEvent</dfn> : Event {
readonly attribute AudioBuffer renderedBuffer;
};
</code></pre>
</div>
</div>
<div id="attributes-OfflineAudioCompletionEvent-section" class="section">
<h3 id="attributes-OfflineAudioCompletionEvent">4.1c.1. Attributes</h3>
<dl>
<dt id="dfn-renderedBuffer"><code>renderedBuffer</code></dt>
<dd><p>An AudioBuffer containing the rendered audio data once an OfflineAudioContext has finished rendering.
It will have a number of channels equal to the <code>numberOfChannels</code> parameter
of the OfflineAudioContext constructor.</p>
</dd>
</dl>
</div>
</div>
<div id="AudioNode-section-section" class="section">
<h2 id="AudioNode-section">4.2. The AudioNode Interface</h2>
<p>AudioNodes are the building blocks of an <a
href="#AudioContext-section"><code>AudioContext</code></a>. This interface
represents audio sources, the audio destination, and intermediate processing
modules. These modules can be connected together to form <a
href="#ModularRouting-section">processing graphs</a> for rendering audio to the
audio hardware. Each node can have <dfn>inputs</dfn> and/or <dfn>outputs</dfn>.
A <dfn>source node</dfn> has no inputs
and a single output. An <a
href="#AudioDestinationNode-section"><code>AudioDestinationNode</code></a> has
one input and no outputs and represents the final destination to the audio
hardware. Most processing nodes such as filters will have one input and one
output. Each type of <code>AudioNode</code> differs in the details of how it processes or synthesizes audio. But, in general, <code>AudioNodes</code>
will process its inputs (if it has any), and generate audio for its outputs (if it has any).
</p>
<p>
Each <dfn>output</dfn> has one or more <dfn>channels</dfn>. The exact number of channels depends on the details of the specific AudioNode.
</p>
<p>
An output may connect to one or more <code>AudioNode</code> inputs, thus <em>fan-out</em> is supported. An input initially has no connections,
but may be connected from one
or more <code>AudioNode</code> outputs, thus <em>fan-in</em> is supported. When the <code>connect()</code> method is called to connect
an output of an AudioNode to an input of an AudioNode, we call that a <dfn>connection</dfn> to the input.
</p>
<p>
Each AudioNode <dfn>input</dfn> has a specific number of channels at any given time. This number can change depending on the <dfn>connection(s)</dfn>
made to the input. If the input has no connections then it has one channel which is silent.
</p>
<p>
For each <dfn>input</dfn>, an <code>AudioNode</code> performs a mixing (usually an up-mixing) of all connections to that input.
Please see <a href="#MixerGainStructure-section">Mixer Gain Structure</a> for more informative details, and the <a href="#UpMix-section">Channel up-mixing and down-mixing</a>
section for normative requirements.
</p>
<p>
For performance reasons, practical implementations will need to use block processing, with each <code>AudioNode</code> processing a
fixed number of sample-frames of size <em>block-size</em>. In order to get uniform behavior across implementations, we will define this
value explicitly. <em>block-size</em> is defined to be 128 sample-frames which corresponds to roughly 3ms at a sample-rate of 44.1KHz.
</p>
<p>
AudioNodes are <em>EventTarget</em>s, as described in <cite><a href="http://dom.spec.whatwg.org/">DOM</a></cite>
<a href="#DOM">[DOM]</a>. This means that it is possible to dispatch events to AudioNodes the same
way that other EventTargets accept events.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-node-idl">
enum <dfn>ChannelCountMode</dfn> {
"max",
"clamped-max",
"explicit"
};
enum <dfn>ChannelInterpretation</dfn> {
"speakers",
"discrete"
};
interface <dfn id="dfn-AudioNode">AudioNode</dfn> : EventTarget {
void connect(AudioNode destination, optional unsigned long output = 0, optional unsigned long input = 0);
void connect(AudioParam destination, optional unsigned long output = 0);
void disconnect(optional unsigned long output = 0);
readonly attribute AudioContext context;
readonly attribute unsigned long numberOfInputs;
readonly attribute unsigned long numberOfOutputs;
// Channel up-mixing and down-mixing rules for all inputs.
attribute unsigned long channelCount;
attribute ChannelCountMode channelCountMode;
attribute ChannelInterpretation channelInterpretation;
};
</code></pre>
</div>
</div>
<div id="attributes-AudioNode-section" class="section">
<h3 id="attributes-AudioNode">4.2.1. Attributes</h3>
<dl>
<dt id="dfn-context"><code>context</code></dt>
<dd><p>The AudioContext which owns this AudioNode.</p>
</dd>
</dl>
<dl>
<dt id="dfn-numberOfInputs_2"><code>numberOfInputs</code></dt>
<dd><p>The number of inputs feeding into the AudioNode. For <dfn>source nodes</dfn>,
this will be 0.</p>
</dd>
</dl>
<dl>
<dt id="dfn-numberOfOutputs_2"><code>numberOfOutputs</code></dt>
<dd><p>The number of outputs coming out of the AudioNode. This will be 0
for an AudioDestinationNode.</p>
</dd>
</dl>
<dl>
<dt id="dfn-channelCount"><code>channelCount</code><dt>
<dd><p>The number of channels used when up-mixing and down-mixing connections to any inputs to the node. The default value is 2
except for specific nodes where its value is specially determined.
This attribute has no effect for nodes with no inputs.
If this value is set to zero, the implementation MUST raise the
NOT_SUPPORTED_ERR exception.</p>
<p>See the <a href="#UpMix-section">Channel up-mixing and down-mixing</a>
section for more information on this attribute.</p>
</dd>
</dl>
<dl>
<dt id="dfn-channelCountMode"><code>channelCountMode</code><dt>
<dd><p>Determines how channels will be counted when up-mixing and down-mixing connections to any inputs to the node
. This attribute has no effect for nodes with no inputs.</p>
<p>See the <a href="#UpMix-section">Channel up-mixing and down-mixing</a>
section for more information on this attribute.</p>
</dd>
</dl>
<dl>
<dt id="dfn-channelInterpretation"><code>channelInterpretation</code><dt>
<dd><p>Determines how individual channels will be treated when up-mixing and down-mixing connections to any inputs to the node.
This attribute has no effect for nodes with no inputs.</p>
<p>See the <a href="#UpMix-section">Channel up-mixing and down-mixing</a>
section for more information on this attribute.</p>
</dd>
</dl>
</div>
<div id="methodsandparams-AudioNode-section" class="section">
<h3 id="methodsandparams-AudioNode">4.2.2. Methods and Parameters</h3>
<dl>
<dt id="dfn-connect-AudioNode">The <code>connect</code> to AudioNode method</dt>
<dd><p>Connects the AudioNode to another AudioNode.</p>
<p>The <dfn id="dfn-destination_2">destination</dfn> parameter is the
AudioNode to connect to.</p>
<p>The <dfn id="dfn-output_2">output</dfn> parameter is an index
describing which output of the AudioNode from which to connect. An
out-of-bound value throws an exception.</p>
<p>The <dfn id="dfn-input_2">input</dfn> parameter is an index describing
which input of the destination AudioNode to connect to. An out-of-bound
value throws an exception. </p>
<p>It is possible to connect an AudioNode output to more than one input
with multiple calls to connect(). Thus, "fan-out" is supported. </p>
<p>
It is possible to connect an AudioNode to another AudioNode which creates a <em>cycle</em>.
In other words, an AudioNode may connect to another AudioNode, which in turn connects back
to the first AudioNode. This is allowed only if there is at least one
<a class="dfnref" href="#DelayNode-section">DelayNode</a> in the <em>cycle</em> or an exception will
be thrown.
</p>
<p>
There can only be one connection between a given output of one specific node and a given input of another specific node.
Multiple connections with the same termini are ignored. For example:
</p>
<pre>
nodeA.connect(nodeB);
nodeA.connect(nodeB);
will have the same effect as
nodeA.connect(nodeB);
</pre>
</dd>
</dl>
<dl>
<dt id="dfn-connect-AudioParam">The <code>connect</code> to AudioParam method</dt>
<dd><p>Connects the AudioNode to an AudioParam, controlling the parameter
value with an audio-rate signal.
</p>
<p>The <dfn id="dfn-destination_3">destination</dfn> parameter is the
AudioParam to connect to.</p>
<p>The <dfn id="dfn-output_3-destination">output</dfn> parameter is an index
describing which output of the AudioNode from which to connect. An
out-of-bound value throws an exception.</p>
<p>It is possible to connect an AudioNode output to more than one AudioParam
with multiple calls to connect(). Thus, "fan-out" is supported. </p>
<p>It is possible to connect more than one AudioNode output to a single AudioParam
with multiple calls to connect(). Thus, "fan-in" is supported. </p>
<p>An AudioParam will take the rendered audio data from any AudioNode output connected to it and <a href="#down-mix">convert it to mono</a> by down-mixing if it is not
already mono, then mix it together with other such outputs and finally will mix with the <em>intrinsic</em>
parameter value (the value the AudioParam would normally have without any audio connections), including any timeline changes
scheduled for the parameter. </p>
<p>
There can only be one connection between a given output of one specific node and a specific AudioParam.
Multiple connections with the same termini are ignored. For example:
</p>
<pre>
nodeA.connect(param);
nodeA.connect(param);
will have the same effect as
nodeA.connect(param);
</pre>
</dd>
</dl>
<dl>
<dt id="dfn-disconnect">The <code>disconnect</code> method</dt>
<dd><p>Disconnects an AudioNode's output.</p>
<p>The <dfn id="dfn-output_3-disconnect">output</dfn> parameter is an index
describing which output of the AudioNode to disconnect. An out-of-bound
value throws an exception.</p>
</dd>
</dl>
</div>
</div>
<h3 id="lifetime-AudioNode">4.2.3. Lifetime</h3>
<p class="norm">This section is informative.</p>
<p>An implementation may choose any method to avoid unnecessary resource usage and unbounded memory growth of unused/finished
nodes. The following is a description to help guide the general expectation of how node lifetime would be managed.
</p>
<p>
An <code>AudioNode</code> will live as long as there are any references to it. There are several types of references:
</p>
<ol>
<li>A <em>normal</em> JavaScript reference obeying normal garbage collection rules. </li>
<li>A <em>playing</em> reference for both <code>AudioBufferSourceNodes</code> and <code>OscillatorNodes</code>.
These nodes maintain a <em>playing</em>
reference to themselves while they are currently playing.</li>
<li>A <em>connection</em> reference which occurs if another <code>AudioNode</code> is connected to it. </li>
<li>A <em>tail-time</em> reference which an <code>AudioNode</code> maintains on itself as long as it has
any internal processing state which has not yet been emitted. For example, a <code>ConvolverNode</code> has
a tail which continues to play even after receiving silent input (think about clapping your hands in a large concert
hall and continuing to hear the sound reverberate throughout the hall). Some <code>AudioNodes</code> have this
property. Please see details for specific nodes.</li>
</ol>
<p>
Any <code>AudioNodes</code> which are connected in a cycle <em>and</em> are directly or indirectly connected to the
<code>AudioDestinationNode</code> of the <code>AudioContext</code> will stay alive as long as the <code>AudioContext</code> is alive.
</p>
<p>
When an <code>AudioNode</code> has no references it will be deleted. But before it is deleted, it will disconnect itself
from any other <code>AudioNodes</code> which it is connected to. In this way it releases all connection references (3) it has to other nodes.
</p>
<p>
Regardless of any of the above references, it can be assumed that the <code>AudioNode</code> will be deleted when its <code>AudioContext</code> is deleted.
</p>
<div id="AudioDestinationNode-section" class="section">
<h2 id="AudioDestinationNode">4.4. The AudioDestinationNode Interface</h2>
<p>This is an <a href="#AudioNode-section"><code>AudioNode</code></a>
representing the final audio destination and is what the user will ultimately
hear. It can often be considered as an audio output device which is connected to
speakers. All rendered audio to be heard will be routed to this node, a
"terminal" node in the AudioContext's routing graph. There is only a single
AudioDestinationNode per AudioContext, provided through the
<code>destination</code> attribute of <a
href="#AudioContext-section"><code>AudioContext</code></a>. </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 0
channelCount = 2;
channelCountMode = "explicit";
channelInterpretation = "speakers";
</pre>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-destination-node-idl">
interface <dfn id="dfn-AudioDestinationNode">AudioDestinationNode</dfn> : AudioNode {
readonly attribute unsigned long maxChannelCount;
};
</code></pre>
</div>
</div>
<div id="attributes-AudioDestinationNode-section" class="section">
<h3 id="attributes-AudioDestinationNode">4.4.1. Attributes</h3>
<dl>
<dt id="dfn-maxChannelCount"><code>maxChannelCount</code></dt>
<dd><p>The maximum number of channels that the <code>channelCount</code> attribute can be set to.
An <code>AudioDestinationNode</code> representing the audio hardware end-point (the normal case) can potentially output more than
2 channels of audio if the audio hardware is multi-channel. <code>maxChannelCount</code> is the maximum number of channels that
this hardware is capable of supporting. If this value is 0, then this indicates that <code>channelCount</code> may not be
changed. This will be the case for an <code>AudioDestinationNode</code> in an <code>OfflineAudioContext</code> and also for
basic implementations with hardware support for stereo output only.</p>
<p><code>channelCount</code> defaults to 2 for a destination in a normal AudioContext, and may be set to any non-zero value less than or equal
to <code>maxChannelCount</code>. An exception will be thrown if this value is not within the valid range. Giving a concrete example, if
the audio hardware supports 8-channel output, then we may set <code>numberOfChannels</code> to 8, and render 8-channels of output.
</p>
<p>
For an AudioDestinationNode in an OfflineAudioContext, the <code>channelCount</code> is determined when the offline context is created and this value
may not be changed.
</p>
</dd>
</dl>
</div>
</div>
<div id="AudioParam-section" class="section">
<h2 id="AudioParam">4.5. The AudioParam Interface</h2>
<p>AudioParam controls an individual aspect of an <a
href="#AudioNode-section"><code>AudioNode</code></a>'s functioning, such as
volume. The parameter can be set immediately to a particular value using the
"value" attribute. Or, value changes can be scheduled to happen at
very precise times (in the coordinate system of AudioContext.currentTime), for envelopes, volume fades, LFOs, filter sweeps, grain
windows, etc. In this way, arbitrary timeline-based automation curves can be
set on any AudioParam. Additionally, audio signals from the outputs of <code>AudioNodes</code> can be connected
to an <code>AudioParam</code>, summing with the <em>intrinsic</em> parameter value.
</p>
<p>
Some synthesis and processing <code>AudioNodes</code> have <code>AudioParams</code> as attributes whose values must
be taken into account on a per-audio-sample basis.
For other <code>AudioParams</code>, sample-accuracy is not important and the value changes can be sampled more coarsely.
Each individual <code>AudioParam</code> will specify that it is either an <em>a-rate</em> parameter
which means that its values must be taken into account on a per-audio-sample basis, or it is a <em>k-rate</em> parameter.
</p>
<p>
Implementations must use block processing, with each <code>AudioNode</code>
processing 128 sample-frames in each block.
</p>
<p>
For each 128 sample-frame block, the value of a <em>k-rate</em> parameter must
be sampled at the time of the very first sample-frame, and that value must be
used for the entire block. <em>a-rate</em> parameters must be sampled for each
sample-frame of the block.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-param-idl">
interface <dfn id="dfn-AudioParam">AudioParam</dfn> {
attribute float value;
readonly attribute float defaultValue;
<span class="comment">// Parameter automation. </span>
void setValueAtTime(float value, double startTime);
void linearRampToValueAtTime(float value, double endTime);
void exponentialRampToValueAtTime(float value, double endTime);
<span class="comment">// Exponentially approach the target value with a rate having the given time constant. </span>
void setTargetAtTime(float target, double startTime, double timeConstant);
<span class="comment">// Sets an array of arbitrary parameter values starting at time for the given duration. </span>
<span class="comment">// The number of values will be scaled to fit into the desired duration. </span>
void setValueCurveAtTime(Float32Array values, double startTime, double duration);
<span class="comment">// Cancels all scheduled parameter changes with times greater than or equal to startTime. </span>
void cancelScheduledValues(double startTime);
};
</code></pre>
</div>
</div>
<div id="attributes-AudioParam-section" class="section">
<h3 id="attributes-AudioParam">4.5.1. Attributes</h3>
<dl>
<dt id="dfn-value"><code>value</code></dt>
<dd><p>The parameter's floating-point value. This attribute is initialized to the
<code>defaultValue</code>. If a value is set during a time when there are any automation events scheduled then
it will be ignored and no exception will be thrown.</p>
</dd>
</dl>
<dl>
<dt id="dfn-defaultValue"><code>defaultValue</code></dt>
<dd><p>Initial value for the value attribute</p>
</dd>
</dl>
</div>
<div id="methodsandparams-AudioParam-section" class="section">
<h3 id="methodsandparams-AudioParam">4.5.2. Methods and Parameters</h3>
<p>
An <code>AudioParam</code> maintains a time-ordered event list which is initially empty. The times are in
the time coordinate system of AudioContext.currentTime. The events define a mapping from time to value. The following methods
can change the event list by adding a new event into the list of a type specific to the method. Each event
has a time associated with it, and the events will always be kept in time-order in the list. These
methods will be called <em>automation</em> methods:</p>
<ul>
<li>setValueAtTime() - <em>SetValue</em></li>
<li>linearRampToValueAtTime() - <em>LinearRampToValue</em></li>
<li>exponentialRampToValueAtTime() - <em>ExponentialRampToValue</em></li>
<li>setTargetAtTime() - <em>SetTarget</em></li>
<li>setValueCurveAtTime() - <em>SetValueCurve</em></li>
</ul>
<p>
The following rules will apply when calling these methods:
</p>
<ul>
<li>If one of these events is added at a time where there is already an event of the exact same type, then the new event will replace the old
one.</li>
<li>If one of these events is added at a time where there is already one or more events of a different type, then it will be
placed in the list after them, but before events whose times are after the event. </li>
<li>If setValueCurveAtTime() is called for time T and duration D and there are any events having a time greater than T, but less than
T + D, then an exception will be thrown. In other words, it's not ok to schedule a value curve during a time period containing other events.</li>
<li>Similarly an exception will be thrown if any <em>automation</em> method is called at a time which is inside of the time interval
of a <em>SetValueCurve</em> event at time T and duration D.</li>
</ul>
<p>
</p>
<dl>
<dt id="dfn-setValueAtTime">The <code>setValueAtTime</code> method</dt>
<dd><p>Schedules a parameter value change at the given time.</p>
<p>The <dfn id="dfn-value_2">value</dfn> parameter is the value the
parameter will change to at the given time.</p>
<p>The <dfn id="dfn-startTime_2">startTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p>
<p>
If there are no more events after this <em>SetValue</em> event, then for t >= startTime, v(t) = value. In other words, the value will remain constant.
</p>
<p>
If the next event (having time T1) after this <em>SetValue</em> event is not of type <em>LinearRampToValue</em> or <em>ExponentialRampToValue</em>,
then, for t: startTime &lt;= t &lt; T1, v(t) = value.
In other words, the value will remain constant during this time interval, allowing the creation of "step" functions.
</p>
<p>
If the next event after this <em>SetValue</em> event is of type <em>LinearRampToValue</em> or <em>ExponentialRampToValue</em> then please
see details below.
</p>
</dd>
</dl>
<dl>
<dt id="dfn-linearRampToValueAtTime">The <code>linearRampToValueAtTime</code>
method</dt>
<dd><p>Schedules a linear continuous change in parameter value from the
previous scheduled parameter value to the given value.</p>
<p>The <dfn id="dfn-value_3">value</dfn> parameter is the value the
parameter will linearly ramp to at the given time.</p>
<p>The <dfn id="dfn-endTime_3">endTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p>
<p>
The value during the time interval T0 &lt;= t &lt; T1 (where T0 is the time of the previous event and T1 is the endTime parameter passed into this method)
will be calculated as:
</p>
<pre>
v(t) = V0 + (V1 - V0) * ((t - T0) / (T1 - T0))
</pre>
<p>
Where V0 is the value at the time T0 and V1 is the value parameter passed into this method.
</p>
<p>
If there are no more events after this LinearRampToValue event then for t >= T1, v(t) = V1
</p>
</dd>
</dl>
<dl>
<dt id="dfn-exponentialRampToValueAtTime">The
<code>exponentialRampToValueAtTime</code> method</dt>
<dd><p>Schedules an exponential continuous change in parameter value from
the previous scheduled parameter value to the given value. Parameters
representing filter frequencies and playback rate are best changed
exponentially because of the way humans perceive sound. </p>
<p>The <dfn id="dfn-value_4">value</dfn> parameter is the value the
parameter will exponentially ramp to at the given time. An exception will be thrown if this value is less than
or equal to 0, or if the value at the time of the previous event is less than or equal to 0.</p>
<p>The <dfn id="dfn-endTime_4">endTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p>
<p>
The value during the time interval T0 &lt;= t &lt; T1 (where T0 is the time of the previous event and T1 is the endTime parameter passed into this method)
will be calculated as:
</p>
<pre>
v(t) = V0 * (V1 / V0) ^ ((t - T0) / (T1 - T0))
</pre>
<p>
Where V0 is the value at the time T0 and V1 is the value parameter passed into this method.
</p>
<p>
If there are no more events after this ExponentialRampToValue event then for t >= T1, v(t) = V1
</p>
</dd>
</dl>
<dl>
<dt id="dfn-setTargetAtTime">The <code>setTargetAtTime</code>
method</dt>
<dd><p>Start exponentially approaching the target value at the given time
with a rate having the given time constant. Among other uses, this is
useful for implementing the "decay" and "release" portions of an ADSR
envelope. Please note that the parameter value does not immediately
change to the target value at the given time, but instead gradually
changes to the target value.</p>
<p>The <dfn id="dfn-target">target</dfn> parameter is the value
the parameter will <em>start</em> changing to at the given time.</p>
<p>The <dfn id="dfn-startTime">startTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p>
<p>The <dfn id="dfn-timeConstant">timeConstant</dfn> parameter is the
time-constant value of first-order filter (exponential) approach to the
target value. The larger this value is, the slower the transition will
be.</p>
<p>
More precisely, <em>timeConstant</em> is the time it takes a first-order linear continuous time-invariant system
to reach the value 1 - 1/e (around 63.2%) given a step input response (transition from 0 to 1 value).
</p>
<p>
During the time interval: <em>T0</em> &lt;= t &lt; <em>T1</em>, where T0 is the <em>startTime</em> parameter and T1 represents the time of the event following this
event (or <em>infinity</em> if there are no following events):
</p>
<pre>
v(t) = V1 + (V0 - V1) * exp(-(t - T0) / <em>timeConstant</em>)
</pre>
<p>
Where V0 is the initial value (the .value attribute) at T0 (the <em>startTime</em> parameter) and V1 is equal to the <em>target</em>
parameter.
</p>
</dd>
</dl>
<dl>
<dt id="dfn-setValueCurveAtTime">The <code>setValueCurveAtTime</code>
method</dt>
<dd><p>Sets an array of arbitrary parameter values starting at the given
time for the given duration. The number of values will be scaled to fit
into the desired duration. </p>
<p>The <dfn id="dfn-values">values</dfn> parameter is a Float32Array
representing a parameter value curve. These values will apply starting at
the given time and lasting for the given duration. </p>
<p>The <dfn id="dfn-startTime_5">startTime</dfn> parameter is the time in the same time coordinate system as AudioContext.currentTime.</p>
<p>The <dfn id="dfn-duration_5">duration</dfn> parameter is the
amount of time in seconds (after the <em>time</em> parameter) where values will be calculated according to the <em>values</em> parameter..</p>
<p>
During the time interval: <em>startTime</em> &lt;= t &lt; <em>startTime</em> + <em>duration</em>, values will be calculated:
</p>
<pre>
v(t) = values[N * (t - startTime) / duration], where <em>N</em> is the length of the <em>values</em> array.
</pre>
<p>
After the end of the curve time interval (t >= <em>startTime</em> + <em>duration</em>), the value will remain constant at the final curve value,
until there is another automation event (if any).
</p>
</dd>
</dl>
<dl>
<dt id="dfn-cancelScheduledValues">The <code>cancelScheduledValues</code>
method</dt>
<dd><p>Cancels all scheduled parameter changes with times greater than or
equal to startTime.</p>
<p>The <dfn>startTime</dfn> parameter is the starting
time at and after which any previously scheduled parameter changes will
be cancelled. It is a time in the same time coordinate system as AudioContext.currentTime.</p>
</dd>
</dl>
</div>
</div>
<div id="computedValue-AudioParam-section" class="section">
<h3>4.5.3. Computation of Value</h3>
<p>
<dfn>computedValue</dfn> is the final value controlling the audio DSP and is computed by the audio rendering thread during each rendering time quantum.
It must be internally computed as follows:
</p>
<ol>
<li>An <em>intrinsic</em> parameter value will be calculated at each time, which is either the value set directly to the .value attribute,
or, if there are any scheduled parameter changes (automation events) with times before or at this time,
the value as calculated from these events. If the .value attribute
is set after any automation events have been scheduled, then these events will be removed. When read, the .value attribute
always returns the <em>intrinsic</em> value for the current time. If automation events are removed from a given time range, then the
<em>intrinsic</em> value will remain unchanged and stay at its previous value until either the .value attribute is directly set, or automation events are added
for the time range.
</li>
<li>
An AudioParam will take the rendered audio data from any AudioNode output connected to it and <a href="#down-mix">convert it to mono</a> by down-mixing if it is not
already mono, then mix it together with other such outputs. If there are no AudioNodes connected to it, then this value is 0, having no
effect on the <em>computedValue</em>.
</li>
<li>
The <em>computedValue</em> is the sum of the <em>intrinsic</em> value and the value calculated from (2).
</li>
</ol>
</div>
<div id="example1-AudioParam-section" class="section">
<h3 id="example1-AudioParam">4.5.4. AudioParam Automation Example</h3>
<div class="example">
<div class="exampleHeader">
Example</div>
<img alt="AudioParam automation" src="images/audioparam-automation1.png" />
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">ECMAScript</span></div>
<div class="blockContent">
<pre class="code"><code class="es-code">
var t0 = 0;
var t1 = 0.1;
var t2 = 0.2;
var t3 = 0.3;
var t4 = 0.4;
var t5 = 0.6;
var t6 = 0.7;
var t7 = 1.0;
var curveLength = 44100;
var curve = new Float32Array(curveLength);
for (var i = 0; i &lt; curveLength; ++i)
curve[i] = Math.sin(Math.PI * i / curveLength);
param.setValueAtTime(0.2, t0);
param.setValueAtTime(0.3, t1);
param.setValueAtTime(0.4, t2);
param.linearRampToValueAtTime(1, t3);
param.linearRampToValueAtTime(0.15, t4);
param.exponentialRampToValueAtTime(0.75, t5);
param.exponentialRampToValueAtTime(0.05, t6);
param.setValueCurveAtTime(curve, t6, t7 - t6);
</code></pre>
</div>
</div>
</div>
</div>
<div id="GainNode-section" class="section">
<h2 id="GainNode">4.7. The GainNode Interface</h2>
<p>Changing the gain of an audio signal is a fundamental operation in audio
applications. The <code>GainNode</code> is one of the building blocks for creating <a
href="#MixerGainStructure-section">mixers</a>.
This interface is an AudioNode with a single input and single
output: </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCountMode = "max";
channelInterpretation = "speakers";
</pre>
<p>It multiplies the input audio signal by the (possibly time-varying) <code>gain</code> attribute, copying the result to the output.
By default, it will take the input and pass it through to the output unchanged, which represents a constant gain change
of 1.
</p>
<p>
As with other <code>AudioParams</code>, the <code>gain</code> parameter represents a mapping from time
(in the coordinate system of AudioContext.currentTime) to floating-point value.
Every PCM audio sample in the input is multiplied by the <code>gain</code> parameter's value for the specific time
corresponding to that audio sample. This multiplied value represents the PCM audio sample for the output.
</p>
<p>
The number of channels of the output will always equal the number of channels of the input, with each channel
of the input being multiplied by the <code>gain</code> values and being copied into the corresponding channel
of the output.
</p>
<p>
The implementation must make
gain changes to the audio stream smoothly, without introducing noticeable
clicks or glitches. This process is called "de-zippering". </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="gain-node-idl">
interface <dfn id="dfn-GainNode">GainNode</dfn> : AudioNode {
readonly attribute AudioParam gain;
};
</code></pre>
</div>
</div>
<div id="attributes-GainNode-section" class="section">
<h3 id="attributes-GainNode">4.7.1. Attributes</h3>
<dl>
<dt id="dfn-gain"><code>gain</code></dt>
<dd><p>Represents the amount of gain to apply. Its
default <code>value</code> is 1 (no gain change). The nominal <code>minValue</code> is 0, but may be
set negative for phase inversion. The nominal <code>maxValue</code> is 1, but higher values are allowed (no
exception thrown).This parameter is <em>a-rate</em> </p>
</dd>
</dl>
</div>
</div>
<div id="DelayNode-section" class="section">
<h2 id="DelayNode">4.8. The DelayNode Interface</h2>
<p>A delay-line is a fundamental building block in audio applications. This
interface is an AudioNode with a single input and single output: </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCountMode = "max";
channelInterpretation = "speakers";
</pre>
<p>
The number of channels of the output always equals the number of channels of the input.
</p>
<p>It delays the incoming audio signal by a certain amount. The default
amount is 0 seconds (no delay). When the delay time is changed, the
implementation must make the transition smoothly, without introducing
noticeable clicks or glitches to the audio stream. </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="delay-node-idl">
interface <dfn id="dfn-DelayNode">DelayNode</dfn> : AudioNode {
readonly attribute AudioParam delayTime;
};
</code></pre>
</div>
</div>
<div id="attributes-GainNode-section_2" class="section">
<h3 id="attributes-GainNode_2">4.8.1. Attributes</h3>
<dl>
<dt id="dfn-delayTime_2"><code>delayTime</code></dt>
<dd><p>An AudioParam object representing the amount of delay (in seconds)
to apply. The default value (<code>delayTime.value</code>) is 0 (no
delay). The minimum value is 0 and the maximum value is determined by the <em>maxDelayTime</em>
argument to the <code>AudioContext</code> method <code>createDelay</code>. This parameter is <em>a-rate</em></p>
</dd>
</dl>
</div>
</div>
<div id="AudioBuffer-section" class="section">
<h2 id="AudioBuffer">4.9. The AudioBuffer Interface</h2>
<p>This interface represents a memory-resident audio asset (for one-shot sounds
and other short audio clips). Its format is non-interleaved IEEE 32-bit linear PCM with a
nominal range of -1 -&gt; +1. It can contain one or more channels. Typically, it would be expected that the length
of the PCM data would be fairly short (usually somewhat less than a minute).
For longer sounds, such as music soundtracks, streaming should be used with the
<code>audio</code> element and <code>MediaElementAudioSourceNode</code>. </p>
<p>
An AudioBuffer may be used by one or more AudioContexts.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-buffer-idl">
interface <dfn id="dfn-AudioBuffer">AudioBuffer</dfn> {
readonly attribute float sampleRate;
readonly attribute long length;
<span class="comment">// in seconds </span>
readonly attribute double duration;
readonly attribute long numberOfChannels;
Float32Array getChannelData(unsigned long channel);
};
</code></pre>
</div>
</div>
<div id="attributes-AudioBuffer-section" class="section">
<h3 id="attributes-AudioBuffer">4.9.1. Attributes</h3>
<dl>
<dt id="dfn-sampleRate_AudioBuffer"><code>sampleRate</code></dt>
<dd><p>The sample-rate for the PCM audio data in samples per second.</p>
</dd>
</dl>
<dl>
<dt id="dfn-length_AudioBuffer"><code>length</code></dt>
<dd><p>Length of the PCM audio data in sample-frames.</p>
</dd>
</dl>
<dl>
<dt id="dfn-duration_AudioBuffer"><code>duration</code></dt>
<dd><p>Duration of the PCM audio data in seconds.</p>
</dd>
</dl>
<dl>
<dt id="dfn-numberOfChannels_AudioBuffer"><code>numberOfChannels</code></dt>
<dd><p>The number of discrete audio channels.</p>
</dd>
</dl>
</div>
<div id="methodsandparams-AudioBuffer-section" class="section">
<h3 id="methodsandparams-AudioBuffer">4.9.2. Methods and Parameters</h3>
<dl>
<dt id="dfn-getChannelData">The <code>getChannelData</code> method</dt>
<dd><p>Returns the <code>Float32Array</code> representing the PCM audio data for the specific channel.</p>
<p>The <dfn id="dfn-channel">channel</dfn> parameter is an index
representing the particular channel to get data for. An index value of 0 represents
the first channel. This index value MUST be less than <code>numberOfChannels</code>
or an exception will be thrown.</p>
</dd>
</dl>
</div>
</div>
<div id="AudioBufferSourceNode-section" class="section">
<h2 id="AudioBufferSourceNode">4.10. The AudioBufferSourceNode Interface</h2>
<p>This interface represents an audio source from an in-memory audio asset in
an <code>AudioBuffer</code>. It is useful for playing short audio assets
which require a high degree of scheduling flexibility (can playback in
rhythmically perfect ways). The start() method is used to schedule when
sound playback will happen. The playback will stop automatically when
the buffer's audio data has been completely
played (if the <code>loop</code> attribute is false), or when the stop()
method has been called and the specified time has been reached. Please see more
details in the start() and stop() description. start() and stop() may not be issued
multiple times for a given
AudioBufferSourceNode. </p>
<pre> numberOfInputs : 0
numberOfOutputs : 1
</pre>
<p>
The number of channels of the output always equals the number of channels of the AudioBuffer
assigned to the .buffer attribute, or is one channel of silence if .buffer is NULL.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-buffer-source-node-idl">
interface <dfn id="dfn-AudioBufferSourceNode">AudioBufferSourceNode</dfn> : AudioNode {
attribute AudioBuffer? buffer;
readonly attribute AudioParam playbackRate;
attribute boolean loop;
attribute double loopStart;
attribute double loopEnd;
void start(optional double when = 0, optional double offset = 0, optional double duration);
void stop(optional double when = 0);
attribute EventHandler onended;
};
</code></pre>
</div>
</div>
<div id="attributes-AudioBufferSourceNode-section" class="section">
<h3 id="attributes-AudioBufferSourceNode">4.10.1. Attributes</h3>
<dl>
<dt id="dfn-buffer_AudioBufferSourceNode"><code>buffer</code></dt>
<dd><p>Represents the audio asset to be played. </p>
</dd>
</dl>
<dl>
<dt id="dfn-playbackRate_AudioBufferSourceNode"><code>playbackRate</code></dt>
<dd><p>The speed at which to render the audio stream. The default
playbackRate.value is 1. This parameter is <em>a-rate</em> </p>
</dd>
</dl>
<dl>
<dt id="dfn-loop_AudioBufferSourceNode"><code>loop</code></dt>
<dd><p>Indicates if the audio data should play in a loop. The default value is false. </p>
</dd>
</dl>
<dl>
<dt id="dfn-loopStart_AudioBufferSourceNode"><code>loopStart</code></dt>
<dd><p>An optional value in seconds where looping should begin if the <code>loop</code> attribute is true.
Its default value is 0, and it may usefully be set to any value between 0 and the duration of the buffer.</p>
</dd>
</dl>
<dl>
<dt id="dfn-loopEnd_AudioBufferSourceNode"><code>loopEnd</code></dt>
<dd><p>An optional value in seconds where looping should end if the <code>loop</code> attribute is true.
Its default value is 0, and it may usefully be set to any value between 0 and the duration of the buffer.</p>
</dd>
</dl>
<dl>
<dt id="dfn-onended_AudioBufferSourceNode"><code>onended</code></dt>
<dd><p>A property used to set the <code>EventHandler</code> (described in <cite><a
href="http://www.whatwg.org/specs/web-apps/current-work/#eventhandler">HTML</a></cite>)
for the ended event that is dispatched to <a
href="#AudioBufferSourceNode-section"><code>AudioBufferSourceNode</code></a>
node types. When the playback of the buffer for an <code>AudioBufferSourceNode</code>
is finished, an event of type <code>Event</code> (described in <cite><a
href="http://www.whatwg.org/specs/web-apps/current-work/#event">HTML</a></cite>)
will be dispatched to the event handler. </p>
</dd>
</dl>
</div>
</div>
<div id="methodsandparams-AudioBufferSourceNode-section" class="section">
<h3 id="methodsandparams-AudioBufferSourceNode">4.10.2. Methods and
Parameters</h3>
<dl>
<dt id="dfn-start">The <code>start</code> method</dt>
<dd><p>Schedules a sound to playback at an exact time.</p>
<p>The <dfn id="dfn-when">when</dfn> parameter describes at what time (in
seconds) the sound should start playing. It is in the same
time coordinate system as AudioContext.currentTime. If 0 is passed in for
this value or if the value is less than <b>currentTime</b>, then the
sound will start playing immediately. <code>start</code> may only be called one time
and must be called before <code>stop</code> is called or an exception will be thrown.</p>
<p>The <dfn id="dfn-offset">offset</dfn> parameter describes
the offset time in the buffer (in seconds) where playback will begin. If 0 is passed
in for this value, then playback will start from the beginning of the buffer.</p>
<p>The <dfn id="dfn-duration">duration</dfn> parameter
describes the duration of the portion (in seconds) to be played. If this parameter is not passed,
the duration will be equal to the total duration of the AudioBuffer minus the <code>offset</code> parameter.
Thus if neither <code>offset</code> nor <code>duration</code> are specified then the implied duration is
the total duration of the AudioBuffer.
</p>
</dd>
</dl>
<dl>
<dt id="dfn-stop">The <code>stop</code> method</dt>
<dd><p>Schedules a sound to stop playback at an exact time.</p>
<p>The <dfn id="dfn-when_AudioBufferSourceNode_2">when</dfn> parameter
describes at what time (in seconds) the sound should stop playing.
It is in the same time coordinate system as AudioContext.currentTime.
If 0 is passed in for this value or if the value is less than
<b>currentTime</b>, then the sound will stop playing immediately.
<code>stop</code> must only be called one time and only after a call to <code>start</code> or <code>stop</code>,
or an exception will be thrown.</p>
</dd>
</dl>
</div>
<div id="looping-AudioBufferSourceNode-section" class="section">
<h3 id="looping-AudioBufferSourceNode">4.10.3. Looping</h3>
<p>
If the <code>loop</code> attribute is true when <code>start()</code> is called, then playback will continue indefinitely
until <code>stop()</code> is called and the stop time is reached. We'll call this "loop" mode. Playback always starts at the point in the buffer indicated
by the <code>offset</code> argument of <code>start()</code>, and in <em>loop</em> mode will continue playing until it reaches the <em>actualLoopEnd</em> position
in the buffer (or the end of the buffer), at which point it will wrap back around to the <em>actualLoopStart</em> position in the buffer, and continue
playing according to this pattern.
</p>
<p>
In <em>loop</em> mode then the <em>actual</em> loop points are calculated as follows from the <code>loopStart</code> and <code>loopEnd</code> attributes:
</p>
<blockquote>
<pre>
if ((loopStart || loopEnd) &amp;&amp; loopStart >= 0 &amp;&amp; loopEnd > 0 &amp;&amp; loopStart &lt; loopEnd) {
actualLoopStart = loopStart;
actualLoopEnd = min(loopEnd, buffer.length);
} else {
actualLoopStart = 0;
actualLoopEnd = buffer.length;
}
</pre>
</blockquote>
<p>
Note that the default values for <code>loopStart</code> and <code>loopEnd</code> are both 0, which indicates that looping should occur from the very start
to the very end of the buffer.
</p>
<p>
Please note that as a low-level implementation detail, the AudioBuffer is at a specific sample-rate (usually the same as the AudioContext sample-rate), and
that the loop times (in seconds) must be converted to the appropriate sample-frame positions in the buffer according to this sample-rate.
</p>
</div>
<div id="MediaElementAudioSourceNode-section" class="section">
<h2 id="MediaElementAudioSourceNode">4.11. The MediaElementAudioSourceNode
Interface</h2>
<p>This interface represents an audio source from an <code>audio</code> or
<code>video</code> element. </p>
<pre> numberOfInputs : 0
numberOfOutputs : 1
</pre>
<p>
The number of channels of the output corresponds to the number of channels of the media referenced by the HTMLMediaElement.
Thus, changes to the media element's .src attribute can change the number of channels output by this node.
If the .src attribute is not set, then the number of channels output will be one silent channel.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="media-element-audio-source-node-idl">
interface <dfn id="dfn-MediaElementAudioSourceNode">MediaElementAudioSourceNode</dfn> : AudioNode {
};
</code></pre>
</div>
</div>
</div>
<p>A MediaElementAudioSourceNode
is created given an HTMLMediaElement using the AudioContext <a href="#dfn-createMediaElementSource">createMediaElementSource()</a> method. </p>
<p>
The number of channels of the single output equals the number of channels of the audio referenced by
the HTMLMediaElement passed in as the argument to createMediaElementSource(), or is 1 if the HTMLMediaElement
has no audio.
</p>
<p>
The HTMLMediaElement must behave in an identical fashion after the MediaElementAudioSourceNode has
been created, <em>except</em> that the rendered audio will no longer be heard directly, but instead will be heard
as a consequence of the MediaElementAudioSourceNode being connected through the routing graph. Thus pausing, seeking,
volume, <code>.src</code> attribute changes, and other aspects of the HTMLMediaElement must behave as they normally would
if <em>not</em> used with a MediaElementAudioSourceNode.
</p>
<div class="example">
<div class="exampleHeader">
Example</div>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">ECMAScript</span></div>
<div class="blockContent">
<pre class="code"><code class="es-code">
var mediaElement = document.getElementById('mediaElementID');
var sourceNode = context.createMediaElementSource(mediaElement);
sourceNode.connect(filterNode);
</code></pre>
</div>
</div>
</div>
</div>
<div id="ScriptProcessorNode-section" class="section">
<h2 id="ScriptProcessorNode">4.12. The ScriptProcessorNode Interface</h2>
<p>This interface is an AudioNode which can generate, process, or analyse audio
directly using JavaScript. </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCount = numberOfInputChannels;
channelCountMode = "explicit";
channelInterpretation = "speakers";
</pre>
<p>The ScriptProcessorNode is constructed with a <code>bufferSize</code> which
must be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384.
This value controls how frequently the <code>audioprocess</code> event
is dispatched and how many sample-frames need to be processed each call.
Lower numbers for <code>bufferSize</code> will result in a lower (better) <a
href="#Latency-section">latency</a>. Higher numbers will be necessary to avoid
audio breakup and <a href="#Glitching-section">glitches</a>.
This value will be picked by the implementation if the bufferSize argument
to <code>createScriptProcessor</code> is not passed in, or is set to 0.</p>
<p><code>numberOfInputChannels</code> and <code>numberOfOutputChannels</code>
determine the number of input and output channels. It is invalid for both
<code>numberOfInputChannels</code> and <code>numberOfOutputChannels</code> to
be zero. </p>
<pre> var node = context.createScriptProcessor(bufferSize, numberOfInputChannels, numberOfOutputChannels);
</pre>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="script-processor-node-idl">
interface <dfn id="dfn-ScriptProcessorNode">ScriptProcessorNode</dfn> : AudioNode {
attribute EventHandler onaudioprocess;
readonly attribute long bufferSize;
};
</code></pre>
</div>
</div>
<div id="attributes-ScriptProcessorNode-section" class="section">
<h3 id="attributes-ScriptProcessorNode">4.12.1. Attributes</h3>
<dl>
<dt id="dfn-onaudioprocess"><code>onaudioprocess</code></dt>
<dd><p>A property used to set the <code>EventHandler</code> (described in <cite><a
href="http://www.whatwg.org/specs/web-apps/current-work/#eventhandler">HTML</a></cite>)
for the audioprocess event that is dispatched to <a
href="#ScriptProcessorNode-section"><code>ScriptProcessorNode</code></a>
node types. An event of type <a
href="#AudioProcessingEvent-section"><code>AudioProcessingEvent</code></a>
will be dispatched to the event handler. </p>
</dd>
</dl>
<dl>
<dt id="dfn-bufferSize_ScriptProcessorNode"><code>bufferSize</code></dt>
<dd><p>The size of the buffer (in sample-frames) which needs to be
processed each time <code>onprocessaudio</code> is called. Legal values
are (256, 512, 1024, 2048, 4096, 8192, 16384). </p>
</dd>
</dl>
</div>
</div>
<div id="AudioProcessingEvent-section" class="section">
<h2 id="AudioProcessingEvent">4.13. The AudioProcessingEvent Interface</h2>
<p>This is an <code>Event</code> object which is dispatched to <a
href="#ScriptProcessorNode-section"><code>ScriptProcessorNode</code></a> nodes. </p>
<p>The event handler processes audio from the input (if any) by accessing the
audio data from the <code>inputBuffer</code> attribute. The audio data which is
the result of the processing (or the synthesized data if there are no inputs)
is then placed into the <code>outputBuffer</code>. </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-processing-event-idl">
interface <dfn id="dfn-AudioProcessingEvent">AudioProcessingEvent</dfn> : Event {
readonly attribute double playbackTime;
readonly attribute AudioBuffer inputBuffer;
readonly attribute AudioBuffer outputBuffer;
};
</code></pre>
</div>
</div>
<div id="attributes-AudioProcessingEvent-section" class="section">
<h3 id="attributes-AudioProcessingEvent">4.13.1. Attributes</h3>
<dl>
<dt id="dfn-playbackTime"><code>playbackTime</code></dt>
<dd><p>The time when the audio will be played in the same time coordinate system as AudioContext.currentTime.
<code>playbackTime</code> allows for very tight synchronization between
processing directly in JavaScript with the other events in the context's
rendering graph. </p>
</dd>
</dl>
<dl>
<dt id="dfn-inputBuffer"><code>inputBuffer</code></dt>
<dd><p>An AudioBuffer containing the input audio data. It will have a number of channels equal to the <code>numberOfInputChannels</code> parameter
of the createScriptProcessor() method. This AudioBuffer is only valid while in the scope of the <code>onaudioprocess</code>
function. Its values will be meaningless outside of this scope.</p>
</dd>
</dl>
<dl>
<dt id="dfn-outputBuffer"><code>outputBuffer</code></dt>
<dd><p>An AudioBuffer where the output audio data should be written. It will have a number of channels equal to the
<code>numberOfOutputChannels</code> parameter of the createScriptProcessor() method.
Script code within the scope of the <code>onaudioprocess</code> function is expected to modify the
<code>Float32Array</code> arrays representing channel data in this AudioBuffer.
Any script modifications to this AudioBuffer outside of this scope will not produce any audible effects.</p>
</dd>
</dl>
</div>
</div>
<div id="PannerNode-section" class="section">
<h2 id="PannerNode">4.14. The PannerNode Interface</h2>
<p>This interface represents a processing node which <a
href="#Spatialization-section">positions / spatializes</a> an incoming audio
stream in three-dimensional space. The spatialization is in relation to the <a
href="#AudioContext-section"><code>AudioContext</code></a>'s <a
href="#AudioListener-section"><code>AudioListener</code></a>
(<code>listener</code> attribute). </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCount = 2;
channelCountMode = "clamped-max";
channelInterpretation = "speakers";
</pre>
<p>
The audio stream from the input will be either mono or stereo, depending on the connection(s) to the input.
</p>
<p>
The output of this node is hard-coded to stereo (2 channels) and <em>currently</em> cannot be configured.
</p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="panner-node-idl">
enum <dfn>PanningModelType</dfn> {
"equalpower",
"HRTF"
};
enum <dfn>DistanceModelType</dfn> {
"linear",
"inverse",
"exponential"
};
interface <dfn id="dfn-PannerNode">PannerNode</dfn> : AudioNode {
<span class="comment">// Default for stereo is HRTF </span>
attribute PanningModelType panningModel;
<span class="comment">// Uses a 3D cartesian coordinate system </span>
void setPosition(double x, double y, double z);
void setOrientation(double x, double y, double z);
void setVelocity(double x, double y, double z);
<span class="comment">// Distance model and attributes </span>
attribute DistanceModelType distanceModel;
attribute double refDistance;
attribute double maxDistance;
attribute double rolloffFactor;
<span class="comment">// Directional sound cone </span>
attribute double coneInnerAngle;
attribute double coneOuterAngle;
attribute double coneOuterGain;
};
</code></pre>
</div>
</div>
</div>
<div id="attributes-PannerNode_attributes-section" class="section">
<h3 id="attributes-PannerNode_attributes">4.14.2. Attributes</h3>
<dl>
<dt id="dfn-panningModel"><code>panningModel</code></dt>
<dd><p>Determines which spatialization algorithm will be used to position
the audio in 3D space. The default is "HRTF". </p>
<dl>
<dt id="dfn-EQUALPOWER"><code>"equalpower"</code></dt>
<dd><p>A simple and efficient spatialization algorithm using equal-power
panning. </p>
</dd>
</dl>
<dl>
<dt id="dfn-HRTF"><code>"HRTF"</code></dt>
<dd><p>A higher quality spatialization algorithm using a convolution with
measured impulse responses from human subjects. This panning method
renders stereo output. </p>
</dd>
</dl>
</dd>
</dl>
<dl>
<dt id="dfn-distanceModel"><code>distanceModel</code></dt>
<dd><p>Determines which algorithm will be used to reduce the volume of an
audio source as it moves away from the listener. The default is "inverse".
</p>
<dl>
<dt id="dfn-LINEAR_DISTANCE"><code>"linear"</code></dt>
<dd><p>A linear distance model which calculates <em>distanceGain</em> according to: </p>
<pre>
1 - rolloffFactor * (distance - refDistance) / (maxDistance - refDistance)
</pre>
</dd>
</dl>
<dl>
<dt id="dfn-INVERSE_DISTANCE"><code>"inverse"</code></dt>
<dd><p>An inverse distance model which calculates <em>distanceGain</em> according to: </p>
<pre>
refDistance / (refDistance + rolloffFactor * (distance - refDistance))
</pre>
</dd>
</dl>
<dl>
<dt id="dfn-EXPONENTIAL_DISTANCE"><code>"exponential"</code></dt>
<dd><p>An exponential distance model which calculates <em>distanceGain</em> according to: </p>
<pre>
pow(distance / refDistance, -rolloffFactor)
</pre>
</dd>
</dl>
</dd>
</dl>
<dl>
<dt id="dfn-refDistance"><code>refDistance</code></dt>
<dd><p>A reference distance for reducing volume as source move further from
the listener. The default value is 1. </p>
</dd>
</dl>
<dl>
<dt id="dfn-maxDistance"><code>maxDistance</code></dt>
<dd><p>The maximum distance between source and listener, after which the
volume will not be reduced any further. The default value is 10000. </p>
</dd>
</dl>
<dl>
<dt id="dfn-rolloffFactor"><code>rolloffFactor</code></dt>
<dd><p>Describes how quickly the volume is reduced as source moves away
from listener. The default value is 1. </p>
</dd>
</dl>
<dl>
<dt id="dfn-coneInnerAngle"><code>coneInnerAngle</code></dt>
<dd><p>A parameter for directional audio sources, this is an angle, inside
of which there will be no volume reduction. The default value is 360. </p>
</dd>
</dl>
<dl>
<dt id="dfn-coneOuterAngle"><code>coneOuterAngle</code></dt>
<dd><p>A parameter for directional audio sources, this is an angle, outside
of which the volume will be reduced to a constant value of
<b>coneOuterGain</b>. The default value is 360. </p>
</dd>
</dl>
<dl>
<dt id="dfn-coneOuterGain"><code>coneOuterGain</code></dt>
<dd><p>A parameter for directional audio sources, this is the amount of
volume reduction outside of the <b>coneOuterAngle</b>. The default value is 0. </p>
</dd>
</dl>
</div>
<h3 id="Methods_and_Parameters">4.14.3. Methods and Parameters</h3>
<dl>
<dt id="dfn-setPosition">The <code>setPosition</code> method</dt>
<dd><p>Sets the position of the audio source relative to the
<b>listener</b> attribute. A 3D cartesian coordinate system is used.</p>
<p>The <dfn id="dfn-x">x, y, z</dfn> parameters represent the coordinates
in 3D space. </p>
<p>The default value is (0,0,0)
</p>
</dd>
</dl>
<dl>
<dt id="dfn-setOrientation">The <code>setOrientation</code> method</dt>
<dd><p>Describes which direction the audio source is pointing in the 3D
cartesian coordinate space. Depending on how directional the sound is
(controlled by the <b>cone</b> attributes), a sound pointing away from
the listener can be very quiet or completely silent.</p>
<p>The <dfn id="dfn-x_2">x, y, z</dfn> parameters represent a direction
vector in 3D space. </p>
<p>The default value is (1,0,0)
</p>
</dd>
</dl>
<dl>
<dt id="dfn-setVelocity">The <code>setVelocity</code> method</dt>
<dd><p>Sets the velocity vector of the audio source. This vector controls
both the direction of travel and the speed in 3D space. This velocity
relative to the listener's velocity is used to determine how much doppler
shift (pitch change) to apply. The units used for this vector is <em>meters / second</em>
and is independent of the units used for position and orientation vectors.</p>
<p>The <dfn id="dfn-x_3">x, y, z</dfn> parameters describe a direction
vector indicating direction of travel and intensity. </p>
<p>The default value is (0,0,0)
</p>
</dd>
</dl>
<div id="AudioListener-section" class="section">
<h2 id="AudioListener">4.15. The AudioListener Interface</h2>
<p>This interface represents the position and orientation of the person
listening to the audio scene. All <a
href="#PannerNode-section"><code>PannerNode</code></a> objects
spatialize in relation to the AudioContext's <code>listener</code>. See <a
href="#Spatialization-section">this</a> section for more details about
spatialization. </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="audio-listener-idl">
interface <dfn id="dfn-AudioListener">AudioListener</dfn> {
attribute double dopplerFactor;
attribute double speedOfSound;
<span class="comment">// Uses a 3D cartesian coordinate system </span>
void setPosition(double x, double y, double z);
void setOrientation(double x, double y, double z, double xUp, double yUp, double zUp);
void setVelocity(double x, double y, double z);
};
</code></pre>
</div>
</div>
</div>
<div id="attributes-AudioListener-section" class="section">
<h3 id="attributes-AudioListener">4.15.1. Attributes</h3>
<dl>
<dt id="dfn-dopplerFactor"><code>dopplerFactor</code></dt>
<dd><p>A constant used to determine the amount of pitch shift to use when
rendering a doppler effect. The default value is 1. </p>
</dd>
</dl>
<dl>
<dt id="dfn-speedOfSound"><code>speedOfSound</code></dt>
<dd><p>The speed of sound used for calculating doppler shift. The default
value is 343.3. </p>
</dd>
</dl>
</div>
<h3 id="L15842">4.15.2. Methods and Parameters</h3>
<dl>
<dt id="dfn-setPosition_2">The <code>setPosition</code> method</dt>
<dd><p>Sets the position of the listener in a 3D cartesian coordinate
space. <code>PannerNode</code> objects use this position relative to
individual audio sources for spatialization.</p>
<p>The <dfn id="dfn-x_AudioListener">x, y, z</dfn> parameters represent
the coordinates in 3D space. </p>
<p>The default value is (0,0,0)
</p>
</dd>
</dl>
<dl>
<dt id="dfn-setOrientation_2">The <code>setOrientation</code> method</dt>
<dd><p>Describes which direction the listener is pointing in the 3D
cartesian coordinate space. Both a <b>front</b> vector and an <b>up</b>
vector are provided. In simple human terms, the <b>front</b> vector represents which
direction the person's nose is pointing. The <b>up</b> vector represents the
direction the top of a person's head is pointing. These values are expected to
be linearly independent (at right angles to each other). For normative requirements
of how these values are to be interpreted, see the
<a href="#Spatialization-section">spatialization section</a>.
</p>
<p>The <dfn id="dfn-x_setOrientation">x, y, z</dfn> parameters represent
a <b>front</b> direction vector in 3D space, with the default value being (0,0,-1) </p>
<p>The <dfn id="dfn-x_setOrientation_2">xUp, yUp, zUp</dfn> parameters
represent an <b>up</b> direction vector in 3D space, with the default value being (0,1,0) </p>
</dd>
</dl>
<dl>
<dt id="dfn-setVelocity_4">The <code>setVelocity</code> method</dt>
<dd><p>Sets the velocity vector of the listener. This vector controls both
the direction of travel and the speed in 3D space. This velocity relative to
an audio source's velocity is used to determine how much doppler shift
(pitch change) to apply. The units used for this vector is <em>meters / second</em>
and is independent of the units used for position and orientation vectors.</p>
<p>The <dfn id="dfn-x_setVelocity_5">x, y, z</dfn> parameters describe a
direction vector indicating direction of travel and intensity. </p>
<p>The default value is (0,0,0)
</p>
</dd>
</dl>
<div id="ConvolverNode-section" class="section">
<h2 id="ConvolverNode">4.16. The ConvolverNode Interface</h2>
<p>This interface represents a processing node which applies a <a
href="#Convolution-section">linear convolution effect</a> given an impulse
response. Normative requirements for multi-channel convolution matrixing are described
<a href="#Convolution-reverb-effect">here</a>. </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCount = 2;
channelCountMode = "clamped-max";
channelInterpretation = "speakers";
</pre>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="convolver-node-idl">
interface <dfn id="dfn-ConvolverNode">ConvolverNode</dfn> : AudioNode {
attribute AudioBuffer? buffer;
attribute boolean normalize;
};
</code></pre>
</div>
</div>
</div>
<div id="attributes-ConvolverNode-section" class="section">
<h3 id="attributes-ConvolverNode">4.16.1. Attributes</h3>
<dl>
<dt id="dfn-buffer_ConvolverNode"><code>buffer</code></dt>
<dd><p>A mono, stereo, or 4-channel <code>AudioBuffer</code> containing the (possibly multi-channel) impulse response
used by the ConvolverNode. This <code>AudioBuffer</code> must be of the same sample-rate as the AudioContext or an exception will
be thrown. At the time when this attribute is set, the <em>buffer</em> and the state of the <em>normalize</em>
attribute will be used to configure the ConvolverNode with this impulse response having the given normalization.
The initial value of this attribute is null.</p>
</dd>
</dl>
<dl>
<dt id="dfn-normalize"><code>normalize</code></dt>
<dd><p>Controls whether the impulse response from the buffer will be scaled
by an equal-power normalization when the <code>buffer</code> atttribute
is set. Its default value is <code>true</code> in order to achieve a more
uniform output level from the convolver when loaded with diverse impulse
responses. If <code>normalize</code> is set to <code>false</code>, then
the convolution will be rendered with no pre-processing/scaling of the
impulse response. Changes to this value do not take effect until the next time
the <em>buffer</em> attribute is set. </p>
</dd>
</dl>
<p>
If the <em>normalize</em> attribute is false when the <em>buffer</em> attribute is set then the
ConvolverNode will perform a linear convolution given the exact impulse response contained within the <em>buffer</em>.
</p>
<p>
Otherwise, if the <em>normalize</em> attribute is true when the <em>buffer</em> attribute is set then the
ConvolverNode will first perform a scaled RMS-power analysis of the audio data contained within <em>buffer</em> to calculate a
<em>normalizationScale</em> given this algorithm:
</p>
<div class="block">
<div class="blockTitleDiv">
<div class="blockContent">
<pre class="code"><code class="es-code">
float calculateNormalizationScale(buffer)
{
const float GainCalibration = 0.00125;
const float GainCalibrationSampleRate = 44100;
const float MinPower = 0.000125;
// Normalize by RMS power.
size_t numberOfChannels = buffer->numberOfChannels();
size_t length = buffer->length();
float power = 0;
for (size_t i = 0; i &lt; numberOfChannels; ++i) {
float* sourceP = buffer->channel(i)->data();
float channelPower = 0;
int n = length;
while (n--) {
float sample = *sourceP++;
channelPower += sample * sample;
}
power += channelPower;
}
power = sqrt(power / (numberOfChannels * length));
// Protect against accidental overload.
if (isinf(power) || isnan(power) || power &lt; MinPower)
power = MinPower;
float scale = 1 / power;
// Calibrate to make perceived volume same as unprocessed.
scale *= GainCalibration;
// Scale depends on sample-rate.
if (buffer->sampleRate())
scale *= GainCalibrationSampleRate / buffer->sampleRate();
// True-stereo compensation.
if (buffer->numberOfChannels() == 4)
scale *= 0.5;
return scale;
}
</code></pre>
</div>
</div>
</div>
<p>
During processing, the ConvolverNode will then take this calculated <em>normalizationScale</em> value and multiply it by the result of the linear convolution
resulting from processing the input with the impulse response (represented by the <em>buffer</em>) to produce the
final output. Or any mathematically equivalent operation may be used, such as pre-multiplying the
input by <em>normalizationScale</em>, or pre-multiplying a version of the impulse-response by <em>normalizationScale</em>.
</p>
</div>
<div id="AnalyserNode-section" class="section">
<h2 id="AnalyserNode">4.17. The AnalyserNode Interface</h2>
<p>This interface represents a node which is able to provide real-time
frequency and time-domain <a href="#AnalyserNode">analysis</a>
information. The audio stream will be passed un-processed from input to output.
</p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1 <em>Note that this output may be left unconnected.</em>
channelCount = 1;
channelCountMode = "explicit";
channelInterpretation = "speakers";
</pre>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="analyser-node-idl">
interface <dfn id="dfn-AnalyserNode">AnalyserNode</dfn> : AudioNode {
<span class="comment">// Real-time frequency-domain data </span>
void getFloatFrequencyData(Float32Array array);
void getByteFrequencyData(Uint8Array array);
<span class="comment">// Real-time waveform data </span>
void getByteTimeDomainData(Uint8Array array);
attribute unsigned long fftSize;
readonly attribute unsigned long frequencyBinCount;
attribute double minDecibels;
attribute double maxDecibels;
attribute double smoothingTimeConstant;
};
</code></pre>
</div>
</div>
</div>
<div id="attributes-ConvolverNode-section_2" class="section">
<h3 id="attributes-ConvolverNode_2">4.17.1. Attributes</h3>
<dl>
<dt id="dfn-fftSize"><code>fftSize</code></dt>
<dd><p>The size of the FFT used for frequency-domain analysis. This must be
a non-zero power of two in the range 32 to 2048, otherwise an INDEX_SIZE_ERR exception MUST be thrown.
The default value is 2048.</p>
</dd>
</dl>
<dl>
<dt id="dfn-frequencyBinCount"><code>frequencyBinCount</code></dt>
<dd><p>Half the FFT size. </p>
</dd>
</dl>
<dl>
<dt id="dfn-minDecibels"><code>minDecibels</code></dt>
<dd><p>The minimum power value in the scaling range for the FFT analysis
data for conversion to unsigned byte values.
The default value is -100.
If the value of this attribute is set to a value more than or equal to <code>maxDecibels</code>,
an INDEX_SIZE_ERR exception MUST be thrown.</p>
</dd>
</dl>
<dl>
<dt id="dfn-maxDecibels"><code>maxDecibels</code></dt>
<dd><p>The maximum power value in the scaling range for the FFT analysis
data for conversion to unsigned byte values.
The default value is -30.
If the value of this attribute is set to a value less than or equal to <code>minDecibels</code>,
an INDEX_SIZE_ERR exception MUST be thrown.</p>
</dd>
</dl>
<dl>
<dt id="dfn-smoothingTimeConstant"><code>smoothingTimeConstant</code></dt>
<dd><p>A value from 0 -&gt; 1 where 0 represents no time averaging
with the last analysis frame.
The default value is 0.8.
If the value of this attribute is set to a value less than 0 or more than 1,
an INDEX_SIZE_ERR exception MUST be thrown.</p>
</dd>
</dl>
</div>
<h3 id="methods-and-parameters">4.17.2. Methods and Parameters</h3>
<dl>
<dt id="dfn-getFloatFrequencyData">The <code>getFloatFrequencyData</code>
method</dt>
<dd><p>Copies the current frequency data into the passed floating-point
array. If the array has fewer elements than the frequencyBinCount, the
excess elements will be dropped. If the array has more elements than
the frequencyBinCount, the excess elements will be ignored.</p>
<p>The <dfn id="dfn-array">array</dfn> parameter is where
frequency-domain analysis data will be copied. </p>
</dd>
</dl>
<dl>
<dt id="dfn-getByteFrequencyData">The <code>getByteFrequencyData</code>
method</dt>
<dd><p>Copies the current frequency data into the passed unsigned byte
array. If the array has fewer elements than the frequencyBinCount, the
excess elements will be dropped. If the array has more elements than
the frequencyBinCount, the excess elements will be ignored.</p>
<p>The <dfn id="dfn-array_2">array</dfn> parameter is where
frequency-domain analysis data will be copied. </p>
</dd>
</dl>
<dl>
<dt id="dfn-getByteTimeDomainData">The <code>getByteTimeDomainData</code>
method</dt>
<dd><p>Copies the current time-domain (waveform) data into the passed
unsigned byte array. If the array has fewer elements than the
fftSize, the excess elements will be dropped. If the array has more
elements than fftSize, the excess elements will be ignored.</p>
<p>The <dfn id="dfn-array_3">array</dfn> parameter is where time-domain
analysis data will be copied. </p>
</dd>
</dl>
<div id="ChannelSplitterNode-section" class="section">
<h2 id="ChannelSplitterNode">4.18. The ChannelSplitterNode Interface</h2>
<p>The <code>ChannelSplitterNode</code> is for use in more advanced
applications and would often be used in conjunction with <a
href="#ChannelMergerNode-section"><code>ChannelMergerNode</code></a>. </p>
<pre>
numberOfInputs : 1
numberOfOutputs : Variable N (defaults to 6) // number of "active" (non-silent) outputs is determined by number of channels in the input
channelCountMode = "max";
channelInterpretation = "speakers";
</pre>
<p>This interface represents an AudioNode for accessing the individual channels
of an audio stream in the routing graph. It has a single input, and a number of
"active" outputs which equals the number of channels in the input audio stream.
For example, if a stereo input is connected to an
<code>ChannelSplitterNode</code> then the number of active outputs will be two
(one from the left channel and one from the right). There are always a total
number of N outputs (determined by the <code>numberOfOutputs</code> parameter to the AudioContext method <code>createChannelSplitter()</code>),
The default number is 6 if this value is not provided. Any outputs
which are not "active" will output silence and would typically not be connected
to anything. </p>
<h3 id="example-1">Example:</h3>
<img alt="channel splitter" src="images/channel-splitter.png" />
<p>Please note that in this example, the splitter does <b>not</b> interpret the channel identities (such as left, right, etc.), but
simply splits out channels in the order that they are input.</p>
<p>One application for <code>ChannelSplitterNode</code> is for doing "matrix
mixing" where individual gain control of each channel is desired. </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="channel-splitter-node-idl">
interface <dfn id="dfn-ChannelSplitterNode">ChannelSplitterNode</dfn> : AudioNode {
};
</code></pre>
</div>
</div>
</div>
<div id="ChannelMergerNode-section" class="section">
<h2 id="ChannelMergerNode">4.19. The ChannelMergerNode Interface</h2>
<p>The <code>ChannelMergerNode</code> is for use in more advanced applications
and would often be used in conjunction with <a
href="#ChannelSplitterNode-section"><code>ChannelSplitterNode</code></a>. </p>
<pre>
numberOfInputs : Variable N (default to 6) // number of connected inputs may be less than this
numberOfOutputs : 1
channelCountMode = "max";
channelInterpretation = "speakers";
</pre>
<p>This interface represents an AudioNode for combining channels from multiple
audio streams into a single audio stream. It has a variable number of inputs (defaulting to 6), but not all of them
need be connected. There is a single output whose audio stream has a number of
channels equal to the sum of the numbers of channels of all the connected
inputs. For example, if an <code>ChannelMergerNode</code> has two connected
inputs (both stereo), then the output will be four channels, the first two from
the first input and the second two from the second input. In another example
with two connected inputs (both mono), the output will be two channels
(stereo), with the left channel coming from the first input and the right
channel coming from the second input. </p>
<h3 id="example-2">Example:</h3>
<img alt="channel merger" src="images/channel-merger.png" />
<p>Please note that in this example, the merger does <b>not</b> interpret the channel identities (such as left, right, etc.), but
simply combines channels in the order that they are input.</p>
<p>Be aware that it is possible to connect an <code>ChannelMergerNode</code>
in such a way that it outputs an audio stream with a large number of channels
greater than the maximum supported by the audio hardware. In this case where such an output is connected
to the AudioContext .destination (the audio hardware), then the extra channels will be ignored.
Thus, the <code>ChannelMergerNode</code> should be used in situations where the number
of channels is well understood. </p>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
<div class="blockContent">
<pre class="code"><code class="idl-code" id="channel-merger-node-idl">
interface <dfn id="dfn-ChannelMergerNode">ChannelMergerNode</dfn> : AudioNode {
};
</code></pre>
</div>
</div>
</div>
<div id="DynamicsCompressorNode-section" class="section">
<h2 id="DynamicsCompressorNode">4.20. The DynamicsCompressorNode Interface</h2>
<p>DynamicsCompressorNode is an AudioNode processor implementing a dynamics
compression effect. </p>
<p>Dynamics compression is very commonly used in musical production and game
audio. It lowers the volume of the loudest parts of the signal and raises the
volume of the softest parts. Overall, a louder, richer, and fuller sound can be
achieved. It is especially important in games and musical applications where
large numbers of individual sounds are played simultaneous to control the
overall signal level and help avoid clipping (distorting) the audio output to
the speakers. </p>
<pre>
numberOfInputs : 1
numberOfOutputs : 1
channelCount = 2;
channelCountMode = "explicit";
channelInterpretation = "speakers";
</pre>
<div class="block">
<div class="blockTitleDiv">
<span class="blockTitle">Web IDL</span></div>
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<pre class="code"><code class="idl-code" id="dynamics-compressor-node-idl">
interface <dfn id="dfn-DynamicsCompressorNode">DynamicsCompressorNode</dfn> : AudioNode {
readonly attribute AudioParam threshold; // in Decibels
readonly attribute AudioParam knee; // in Decibels
readonly attribute AudioParam ratio; // unit-less
readonly attribute AudioParam reduction; // in Decibels